2012-08-30 Asterisk Development Team
* Asterisk 10.7.1 Released.
* AST-2012-013: Resolve ACL rules being ignored during calls by some
IAX2 peers
* AST-2012-012: Resolve AMI User Unauthorized Shell Access through
ExternalIVR
2012-07-30 Asterisk Development Team
* Asterisk 10.7.0 Released.
2012-07-11 Asterisk Development Team
* Asterisk 10.7.0-rc1 Released.
2012-07-10 13:35 +0000 [r369871] Kinsey Moore
* main/pbx.c, /, apps/app_stack.c: Improve Goto and GotoIf related
documentation Correct documentation on labeliftrue and
labeliffalse parameters of GotoIf() and update several other
locations that use the same syntax. (closes issue ASTERISK-20007)
Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged
revisions 369869 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-07-09 17:06 +0000 [r369819] Jason Parker
* configs/sip_notify.conf.sample, /: Add Digium phones context to
sip_notify sample config. This makes it so that they can be
reconfigured remotely. (closes issue ASTERISK-19910) ........
Merged revisions 369818 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-07-09 14:43 +0000 [r369793] Jonathan Rose
* /, channels/chan_sip.c: chan_sip: Fix small behavioral change
accidentally introduced in r369750 When removing the warning for
AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed
the return value, which would likely make the indication not be
sent in audio. This fixes that while still removing the warning
message. ........ Merged revisions 369792 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-07-06 21:02 +0000 [r369751] Jonathan Rose
* /, channels/chan_sip.c: chan_sip: Add case for FLASH control
frames so that we don't display a warning. chan_sip channels can
receive flash control frames when connected to analog phones and
possibly for other reasons. There really isn't a reason to warn
when these frames are received, we can safely ignore them.
Patches: dahdi_sip_flash.diff uploaded by Jonathan Rose (license
6182) ........ Merged revisions 369750 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-07-06 18:47 +0000 [r369709-369732] Mark Michelson
* main/tcptls.c, /: Remove a superfluous and dangerous freeing of
an SSL_CTX. The problem here is that multiple server sessions
share a SSL_CTX. When one session ended, the SSL_CTX would be
freed and set NULL, leaving the other sessions unable to
function. The code being removed is superfluous because the
SSL_CTX structures for servers will be properly freed when
ast_ssl_teardown is called. (closes issue ASTERISK-20074)
Reported by Trevor Helmsley Patches: ASTERISK-20074.diff uploaded
by Mark Michelson (license #5049) Testers: Trevor Helmsley
........ Merged revisions 369731 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, main/bridging.c: Fix bridging thread leak. The bridge thread
was exiting but was never being reaped using pthread_join(). This
has been fixed now by calling pthread_join() in
ast_bridge_destroy(). (closes issue ASTERISK-19834) Reported by
Marcus Hunger Review: https://reviewboard.asterisk.org/r/2012
........ Merged revisions 369708 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-07-05 19:12 +0000 [r369653] Kinsey Moore
* apps/app_voicemail.c, /: AST-2012-011: Resolve heap corruption
issue with voicemail The heard and deleted arrays in the
voicemail state structure were not handled properly following the
memory leak fix in r354890 and a fix for an invalid free in
r356797. This could result in accessing and writing into freed
memory. The allocation for these arrays has been reworked to
avoid the possibility of invalid frees, access of freed memory,
and crashes that were occurring as a result of this. Locking
around accesses and modifications of the voicemail state
structure members dh_arraysize, heard, and deleted has been added
to prevent simultaneous modification and access when IMAP storage
is in use. If IMAP storage is not in use, this locking is not
compiled in. Review: https://reviewboard.asterisk.org/r/1994/
(closes issue ASTERISK-19923) Reported by: Dan Delaney Tested by:
Dan Delaney, Julian Yap Patches: vm_alloc_fix.diff uploaded by
kmoore (license 6273) ........ Merged revisions 369652 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-07-05 17:02 +0000 [r369627] Matthew Jordan
* /, channels/chan_sip.c: Do not send a BYE when a provisional
response arrives during a re-INVITE Commits r369557 and r369579
were done to improve handling of re-INVITEs when the UA that was
supposed to receive the re-INVITE fails to respond. A limitation
of those patches occurred when a UA sent a provisional response
to the re-INVITE. This triggered a sending of a BYE in
check_pending. This patch tweaks the handling of the re-INVITE
such that a BYE is not sent in response to those messages. (issue
ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies
patches: (reinvite_tweak.diff license #5012 by Steve Davies)
........ Merged revisions 369626 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-07-03 17:02 +0000 [r369558-369580] Terry Wilson
* /, channels/chan_sip.c: More improvements to re-INVITEs timing
out after a provisional response There is no need to call
check_pendings() on a final response to an INVITE when destroying
the scheduler entry as it will be done later during normal
processing. (issue ASTERISK-19992) ........ Merged revisions
369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_sip.c, channels/sip/include/sip.h: Better handle
re-INVITEs with provisional but no final repsonses A previous
attempt at fixing this issue had negative side effects related to
attended transfers which this patch should resolve. Many thanks
to Steve Davies for all of the good suggestions and testing.
(closes issue ASTERISK-19992) Reported by: Steve Davies Tested
by: Steve Davies, Terry Wilson Review:
https://reviewboard.asterisk.org/r/2009/ ........ Merged
revisions 369557 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-29 20:28 +0000 [r369511] Mark Michelson
* main/rtp_engine.c: Fix apparent copy and paste error where
incorrect "glue" is used.
2012-06-29 16:54 +0000 [r369472-369491] Joshua Colp
* /, channels/chan_sip.c: With some configurations a transport is
not actually specified so assume UDP in these cases. ........
Merged revisions 369490 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_sip.c: Make the address family filter specific
to the transport. (closes issue ASTERISK-16618) Reported by: Leif
Madsen Review: https://reviewboard.asterisk.org/r/1667/ ........
Merged revisions 369471 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-27 21:10 +0000 [r369437] Terry Wilson
* /, channels/chan_sip.c, channels/sip/include/sip.h: AST-2012-010:
Clean up after a reinvite that never gets a final response The
basic problem is that if a re-INVITE is sent by Asterisk and it
receives a provisional response, but no final response, then the
dialog is never torn down. In addition to leaking memory, this
also leaks file descriptors and will eventually lead to Asterisk
no longer being able to process calls. This patch just keeps
track of whether there is an outstanding re-INVITE, and if there
is goes ahead and cleans up everything as though there was no
outstanding reinvite. Review:
https://reviewboard.asterisk.org/r/2009/ (closes issue
ASTERISK-19992) Reported by: Steve Davies Tested by: Steve
Davies, Terry Wilson ........ Merged revisions 369436 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-26 13:22 +0000 [r369369-369391] Matthew Jordan
* /, main/adsi.c: Fix crash in unloading of res_adsi module When
res_adsi is unloaded, it removes the ADSI functions that it
previously installed by passing a NULL adsi_funcs pointer to
ast_adsi_install_funcs. This function was not checking whether or
not the adsi_funcs pointer passed in was NULL before
dereferencing it to check whether or not the version of the
functions matches what the core was expecting it. This patch
makes it so that the version is only checked if a potentially
valid adsi_funcs pointer was passed in. Passing in NULL removes
the installed functions, bypassing the version check. ........
Merged revisions 369390 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* main/cdr.c, /: Fix incorrect duration reporting in CDRs created
in batch mode Certain places in core/cdr.c would, if the duration
value were 0, calculate the duration as being the delta between
the current time and the time at which the CDR record was
started. While this does not typically cause a problem in
non-batch mode, this can cause an issue in batch mode where CDR
records are gathered and written long after those calls have
ended. In particular, this affects calls that were never
answered, as those are expected to have a duration of 0. Often,
this would result in CDR logs with a significant number of calls
with lengthy durations, but dispositions of "BUSY". Note that
this does not affect cdr_csv, as that backend does not use
ast_cdr_getvar and instead directly reports the duration value.
The affected core backends include cdr_apative_odbc and
cdr_custom; other extended or deprecated CDR backends may
potentially still directly manipulate the duration values. (issue
ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883)
Reported by: Thomas Arimont Tested by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1996/ ........ Merged
revisions 369351 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-25 19:16 +0000 [r369353] Mark Michelson
* /, channels/chan_sip.c, channels/sip/include/sip.h: Re-fix how
local tag is generated when sending a 481 to an INVITE. Match our
local tag to whatever to-tag was sent in the initial INVITE.
Because the size of the to-tag may not fit in the buffer in the
sip_pvt, it has been changed to a string field. (closes issue
ASTERISK-19892) reported by Walter Doekes Review:
https://reviewboard.asterisk.org/r/1977 ........ Merged revisions
369352 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-25 15:59 +0000 [r369328] Richard Mudgett
* /, main/features.c: Fix Bridge application occasionally returning
to the wrong location. * Fix do_bridge_masquerade() getting the
resume location from the zombie channel. The code must not touch
a clone channel after it has masqueraded it. The clone channel
has become a zombie and is starting to hangup. (closes issue
ASTERISK-19985) Reported by: jamicque Patches:
jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: jamicque ........ Merged revisions 369327
from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-25 15:52 +0000 [r369303-369325] Mark Michelson
* include/asterisk/adsi.h, /, main/Makefile, res/res_adsi.c,
main/adsi.c (added), res/res_adsi.exports.in (removed): Multiple
revisions 369323-369324 ........ r369323 | mmichelson |
2012-06-25 10:35:43 -0500 (Mon, 25 Jun 2012) | 9 lines Eliminate
embedding of res_adsi.so module. The way this is done is to stop
using the optional API. Instead, res_adsi.so, when loaded fills
in a table of function pointers. Review:
https://reviewboard.asterisk.org/r/1991 ........ r369324 |
mmichelson | 2012-06-25 10:50:17 -0500 (Mon, 25 Jun 2012) | 2
lines Forgot to svn add this file in my last commit. ........
Merged revisions 369323-369324 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_sip.c: Be more consistent with the return code
for requests received from invalid domain. When Asterisk receives
an INVITE from an external domain when allowexternaldomains=no
send a 403 instead of a 404. This is consistent with Asterisk's
behavior when receiving a REGISTER in this situation. (Closes
issue ASTERISK-19601) Reported by Matthew Jordan Patches:
ASTERISK-19601-no401.patch uploaded by Mark Michelson (License
#5049) ........ Merged revisions 369302 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-23 00:12 +0000 [r369236-369283] Richard Mudgett
* /, main/features.c: Fix Bridge application and AMI Bridge action
error handling. * Fix AMI Bridge action disconnecting the AMI
link on error. * Fix AMI Bridge action and Bridge application not
checking if their masquerades were successful. * Fix Bridge
application running the h-exten when it should not. * Made
do_bridge_masquerade() return if the masquerade was successful so
the Bridge application and AMI Bridge action could deal with it
correctly. * Made bridge_call_thread_launch() hangup the passed
in channels if the bridge_call_thread fails to start. Those
channels would have been orphaned. * Made builtin_atxfer() check
the success of the transfer masquerade setup. ........ Merged
revisions 369282 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, apps/app_queue.c: Explicitly check caller hangup in app Queue
rather than a polluted res2 value. ........ Merged revisions
369262 from http://svn.asterisk.org/svn/asterisk/branches/1.8
* apps/app_dial.c, /: Check if PBX was started and fix F and F(x)
action logic in Dial application. ........ Merged revisions
369258 from http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, main/ccss.c: Check if PBX was started for generic CCSS recall.
........ Merged revisions 369238 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_sip.c: Change incorrect chan_sip zombie hangup
debug message. They are all zombies now. ........ Merged
revisions 369235 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-22 19:34 +0000 [r369215] Terry Wilson
* /, channels/chan_sip.c: Don't crash on a guest directmedia call A
sip_pvt may not have relatedpeer set if a call doesn't match up
with a peer. If there is no relatedpeer, there is no direct media
ACL to apply, so just return that it is allowed. (closes issue
ASTERISK-20040) Reported by: Terry Wilson ........ Merged
revisions 369214 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-22 17:23 +0000 [r369206] Kinsey Moore
* /, channels/chan_sip.c: Don't parse media stream state for SIP
video streams The sendonly/recvonly/sendrecv/inactive media
stream attributes were parsed for video, but nothing was ever
done with them. With this code removed, an UNSUPPORTED message is
produced when these attributes are used in conjunction with a
video stream which is the better behavior since they were never
really supported in the first place. ........ Merged revisions
369195 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-20 17:36 +0000 [r369147] Alexandr Anikin
* addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c, /: fix
locking issue on empty callList (issue ASTERISK-19298) Reported
by: Dmitry Melekhov Patches: ASTERISK-18322-2.patch ........
Merged revisions 369146 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-20 02:04 +0000 [r369109] Michael L. Young
* main/netsock2.c, /, include/asterisk/netsock2.h: Fix NULL pointer
segfault in ast_sockaddr_parse() While working with
ast_parse_arg() to perform a validity check, a segfault occurred.
The segfault occurred due to passing a NULL pointer to
ast_sockaddr_parse() from ast_parse_arg(). According to the
documentation in config.h, "result pointer to the result. NULL is
valid here, and can be used to perform only the validity checks."
This patch fixes the segfault by checking for a NULL pointer.
This patch also adds documentation to netsock2.h about why it is
necessary to check for a NULL pointer. (Closes issue
ASTERISK-20006) Reported by: Michael L. Young Tested by: Michael
L. Young Patches: asterisk-20006-netsock-null-ptr.diff uploaded
by Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/1990/ ........ Merged
revisions 369108 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-19 23:32 +0000 [r369091] Alexandr Anikin
* addons/chan_ooh323.c: check rtptimeouts in ooh323 channels as per
config file (rtp voice, video, udptl except rtcp) (closes issue
ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches:
19179-ooh323-ast10.patch
2012-06-19 15:37 +0000 [r369067] Mark Michelson
* /, channels/chan_sip.c: Fix request routing issue when
outboundproxy is used. Asterisk was incorrectly setting the
destination of CANCELs and ACKs for error responses to the URI of
the initial INVITE. This resulted in further requests, such as
INVITEs with authentication credentials, to be routed
incorrectly. Instead, when these CANCEL or ACKs are to be sent,
we should simply keep the destination the same as what it
previously was. There is no need to alter it any. (closes issue
ASTERISK-20008) Reported by Marcus Hunger Patches:
ASTERISK-20008.patch uploaded by Mark Michelson (license #5049)
........ Merged revisions 369066 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-18 18:11 +0000 [r369044] Richard Mudgett
* /, main/features.c: Fix monitoring calls put in a parking lot. *
Fix a regression that was introduced by -r366167 which
effectively disabled monitoring parked calls. (closes issue
ASTERISK-20012) Reported by: sdolloff Tested by: rmudgett
........ Merged revisions 369043 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-15 16:07 +0000 [r369005] Kevin P. Fleming
* channels/sip/sdp_crypto.c, main/slinfactory.c, main/translate.c,
main/jitterbuf.c, main/acl.c, channels/iax2-provision.c,
channels/sig_pri.c, utils/astdb2bdb.c, main/chanvars.c,
main/data.c, main/hashtab.c, channels/chan_misdn.c,
main/abstract_jb.c, main/fixedjitterbuf.c,
channels/sip/dialplan_functions.c, main/test.c, res/snmp/agent.c,
main/event.c, main/astmm.c, channels/sip/config_parser.c,
channels/vgrabbers.c, main/alaw.c, main/asterisk.c, main/dsp.c,
main/timing.c, main/udptl.c, main/autoservice.c,
main/fskmodem_float.c, main/frame.c, main/security_events.c,
main/ccss.c, main/threadstorage.c, main/say.c,
channels/console_video.c, channels/sip/reqresp_parser.c,
main/devicestate.c, main/astfd.c, main/ssl.c,
main/taskprocessor.c, main/autochan.c, channels/misdn/isdn_lib.c,
main/enum.c, main/format_pref.c, main/astobj2.c,
main/indications.c, main/fskmodem.c, channels/misdn_config.c,
apps/confbridge/conf_config_parser.c, main/io.c, main/cli.c,
main/ulaw.c, main/dial.c, main/framehook.c, main/format_cap.c,
main/strcompat.c, main/heap.c, channels/misdn/ie.c, main/plc.c,
main/logger.c, main/stdtime/localtime.c, channels/sig_ss7.c,
main/sched.c, main/datastore.c, main/lock.c, main/strings.c,
main/pbx.c, main/stun.c, channels/sip/srtp.c, main/dnsmgr.c,
channels/vcodecs.c, channels/sip/security_events.c,
utils/astdb2sqlite3.c, main/aoc.c, pbx/dundi-parser.c,
main/cel.c, channels/iax2-parser.c,
build_tools/find_missing_support_level (added), main/netsock.c,
main/tcptls.c, main/srv.c, main/privacy.c, main/callerid.c,
main/file.c, channels/misdn/portinfo.c, main/audiohook.c,
main/xmldoc.c, main/netsock2.c, main/format.c,
main/global_datastores.c, main/rtp_engine.c, /, res/ais/clm.c,
main/utils.c, channels/misdn/isdn_msg_parser.c, main/xml.c,
main/config.c, main/loader.c, main/term.c, main/channel.c,
main/cdr.c, res/ael/pval.c, channels/sig_analog.c, main/tdd.c,
channels/console_gui.c, res/ais/evt.c, main/fskmodem_int.c,
channels/console_board.c, main/syslog.c, main/app.c,
main/image.c, main/dns.c, main/message.c, main/db.c,
main/bridging.c: Multiple revisions 369001-369002 ........
r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun
2012) | 11 lines Add support-level indications to many more
source files. Since we now have tools that scan through the
source tree looking for files with specific support levels, we
need to ensure that every file that is a component of a 'core' or
'extended' module (or the main Asterisk binary) is explicitly
marked with its support level. This patch adds support-level
indications to many more source files in tree, but avoids adding
them to third-party libraries that are included in the tree and
to source files that don't end up involved in Asterisk itself.
........ r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15
Jun 2012) | 3 lines Add a script to enable finding source files
without support-levels defined. ........ Merged revisions
369001-369002 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-14 17:31 +0000 [r368947] Matthew Jordan
* channels/chan_skinny.c: AST-2012-009: Fix crash in chan_skinny
due to Key Pad Button Message handling AST-2012-008 (r367844)
fixed a denial of service attack exploitable in the Skinny
channel driver that occurred when certain messages are sent after
a previously registered station sends an Off Hook message.
Unresolved in that patch is an issue in the Asterisk 10 releases,
wherein, if a Station Key Pad Button Message is processed after
an Off Hook message, the channel driver will inappropriately
dereference a NULL pointer. This patch fixes those places where
the message handling or the channel callback functions would
attempt to dereference the line's pointer to the device. (issue
ASTERISK-19905) Reported by: Christoph Hebeisen Tested by:
mjordan, Christoph Hebeisen Patches: AST-2012-009-10.diff
uploaded by mjordan (license 6283)
2012-06-14 15:25 +0000 [r368899-368928] Mark Michelson
* /, main/Makefile: Revert Makefile change to remove embedding
res_adsi.so The change has resulted in a linking error for
certain versions of GCC. This is much worse than the original
issue, so for now, temporarily revert the change. A more thorough
change will be sought out. ........ Merged revisions 368927 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, funcs/func_volume.c: Fix a deadlock that occurs when
func_volume is used on a local channel. This was discovered by
trying to perform a call forward to an extension that makes use
of func_volume. When the local channel is optimized away, the
datastore on the local;2 channel would have its audiohook
destroyed rather than detaching the audiohook from the channel
and then destroying it. With this patch, func_volume's datastore
destructor takes the proper route of detaching the audiohook and
then destroying it. (closes issue ASTERISK-19611) reported by
Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark
Michelson (license #5049) ........ Merged revisions 368898 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-13 20:27 +0000 [r368895] Matthew Jordan
* res/res_smdi.c, /, res/res_adsi.c: Mark res_smdi/res_adsi as
'core' supported modules Recently, various issues surrounding
weak symbols have caused problems with modules that rely on that
feature to be enabled in menuselect. This includes app_voicemail
and chan_dahdi, as they both rely upon res_smdi and res_adsi,
which, in certain circumstances, may not be enabled by default in
menuselect. Because res_smdi/res_adsi are dependencies for
chan_dahdi/app_voicemail, this patch marks both as 'core'
supported modules. This will allow both app_voicemail and
chan_dahdi to be enabled as well, regardless of whether or not
that system supports weak symbols. (issue AST-900) Reported by:
Thomas Arimont (issue AST-885) Reported by: Denis Alberto
Martinez ........ Merged revisions 368894 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-13 19:36 +0000 [r368885] Mark Michelson
* /, main/Makefile: Remove forced linking of res_adsi.o In GCC 4.5+
the result is that Asterisk has a phantom module loaded at
startup, claiming to be res_adsi. (closes issue ASTERISK-19920)
reported by Leif Madsen ........ Merged revisions 368873 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-13 14:30 +0000 [r368831-368853] Matthew Jordan
* Makefile, /: Do not install empty directories; add ASTLIBDIR
r368830 modified the installation script to only create a
directory if that directory does not exist. If some directory
variable was empty, it would attempt to create the empty
location. It also failed to create the ASTLIBDIR directory. This
patch fixes it such that the correct directories are made and
only created if a value specifying them actually exists. ........
Merged revisions 368852 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* Makefile, /: Do not perform install on existing directories If a
directory already exists, performing a 'make install' will remove
the permissions associated with the current directory and replace
them with the permissions of the user executing the install. This
patch changes this behavior to only perform an install on the
directory if the directory does not exist. Thus, if a user later
changes the permissions on that directory, those permissions will
be preserved in subsequent installs. Review:
https://reviewboard.asterisk.org/r/1986 Review:
https://reviewboard.asterisk.org/r/1864 (closes issue
ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger,
Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded
by mjordan) ........ Merged revisions 368830 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-12 15:37 +0000 [r368808] Mark Michelson
* /, channels/chan_sip.c: Set the Caller ID "tag" on peers even if
remote party information is present. On incoming calls, we were
setting the cid_tag on the dialog only if there was no remote
party information (Remote-Party-ID or P-Asserted-Identity)
present. The Caller ID tag is an invented parameter, though, and
should be set no matter the circumstance. (closes issue
ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884)
Reported by Trey Blancher ........ Merged revisions 368807 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-11 17:08 +0000 [r368760] Richard Mudgett
* main/channel.c, channels/chan_dahdi.c, channels/sig_analog.c, /,
channels/chan_sip.c, include/asterisk/channel.h,
channels/chan_iax2.c: Fix deadlock potential with
ast_set_hangupsource() calls. Calling ast_set_hangupsource() with
the channel lock held can result in a deadlock because the
function also locks the bridged channel. (issue ASTERISK-19537)
(closes issue AST-891) Reported by: Guenther Kelleter Tested by:
Guenther Kelleter (closes issue ASTERISK-19801) Reported by: Alec
Davis ........ Merged revisions 368759 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-11 15:15 +0000 [r368721-368739] Kinsey Moore
* apps/app_voicemail.c, main/udptl.c, channels/sip/sdp_crypto.c, /,
channels/chan_sip.c, main/say.c, res/res_fax.c,
funcs/func_strings.c, channels/sip/reqresp_parser.c,
apps/app_queue.c, main/loader.c, channels/chan_dahdi.c,
res/res_config_odbc.c, channels/sip/dialplan_functions.c,
pbx/pbx_config.c, apps/app_directory.c, res/res_speech.c,
res/res_odbc.c: Fix coverity UNUSED_VALUE findings in core
support level files Most of these were just saving returned
values without using them and in some cases the variable being
saved to could be removed as well. (issue ASTERISK-19672)
........ Merged revisions 368738 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, main/md5.c: Fix compilation in dev-mode Backport a compilation
fix in md5.c from trunk that only showed up in dev-mode under
certain compiler versions. ........ Merged revisions 368719 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-07-10 Asterisk Development Team
* Asterisk 10.6.0 Released.
2012-07-06 Asterisk Development Team
* Asterisk 10.6.0-rc2 Released.
* AST-2012-009: Skinny Channel Driver Remote Crash Vulnerability
* AST-2012-010: Possible Resource Leak on Uncompleted Re-INVITE
Transactions
* AST-2012-011: Remote Crash Vulnerability in VoiceMail Application
* Fix crash on a guest directmedia call
A sip_pvt may not have relatedpeer set if a call doesn't match up
with a peer. If there is no relatedpeer, there is no direct media
ACL to apply, so just return that is is allowed.
(closes issue ASTERISK-20040)
* Fix request routing issue when outboundproxy is used
Asterisk was incorrectly setting the destination of CANCELs and ACKs
for error responses to the URI of the initial INVITE. This resulted
in further requests, such as INVITEs with authentication
credentials, to be routed incorrectly. Instead when these CANCEL or
ACKs are to be esnt, we should simply keep the destination the same
as what it previously was. There is no need to alter it any.
(closes issue ASTERISK-20008)
* Fix monitoring calls put in a parking lot
Fix a regression that was introduced by r366167 which effectively
disabled monitoring parked calls.
(closes issue ASTERISK-20012)
2012-06-08 Asterisk Development Team
* Asterisk 10.6.0-rc1 Released.
2012-06-06 21:32 +0000 [r368645] Richard Mudgett
* channels/chan_dahdi.c, channels/sig_analog.c, /: Fix POTS flash
hook to orignate a second call deadlock. A deadlock can occur
when a POTS phone tries to flash hook to originate a second call
for 3-way or transfer. If another process is scanning the
channels container when the POTS line flash hooks then a deadlock
will occur. * Release the channel and private locks when creating
a new channel as a result of a flash hook. (closes issue
ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett
........ Merged revisions 368644 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-06 19:18 +0000 [r368629] Mark Michelson
* /, channels/chan_sip.c: Fix a specific scenario where ACKs are
not matched. If a dialog-starting INVITE contains a to-tag, then
Asterisk will respond with a 481. In this case, the resulting
incoming ACK would not be matched, so Asterisk would continue
retransmitting the 481 until the transaction times out. There
were two issues. Asterisk, upon creating a sip_pvt would generate
a local tag. However, when the time came to transmit the 481,
since there was a to-tag in the INVITE, Asterisk would place this
original to-tag in the 481 response. When the ACK came in,
Asterisk would attempt to match the to-tag in the ACK to the
generated local tag. Unfortunately, Asterisk never actually
transmitted a response with the generated local tag, so the
to-tag in the ACK would not match. The other problem was that
when the 481 was sent, nothing was set on the sip_pvt to indicate
what CSeq is expected in the ACK. To fix the first problem, we
zero out the to-tag seen in the incoming INVITE. This way,
Asterisk, when time to send a response, will send its generated
local tag instead. To fix the second problem, we set the
sip_pvt's pendinginvite to the CSeq of the INVITE when we send a
481. (closes issue ASTERISK-19892) Reported by Mark Michelson
........ Merged revisions 368625 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-06 17:21 +0000 [r368605] Matthew Jordan
* /, build_tools/make_version: Add feature modifier to versions
produced from branches Certain branches, such as Certified
Asterisk, may have a modifier added to them that specifies the
features available in that branch. For branches, this modifier is
expected to be reflected in the location of the branch in
subversion. For example, a subversion of URL of
/certified/branches/1.8.11 would have a feature modifier of
'certified'. This is slightly different then how features are
determined for tags, where the feature is part of the actual tag
name, e.g., "10.5.0-digiumphones". In keeping with the
nomenclature used for tags, the feature specifier for branches is
translated and placed after the revision numbers. For the example
given previously, this would result in a branch version of
"Asterisk SVN-branch-1.8.11-cert-rXXXXXX". ........ Merged
revisions 368604 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-06 16:09 +0000 [r368587] Kinsey Moore
* /, channels/chan_sip.c: Ensure overlapping hold flags do not
conflict When changing between different modes of hold, the flags
were not being cleared out properly causing a failure to change
hold states. (closes issue ASTERISK-19919) Patch-by: Morten
Tryfoss Reported-by: Morten Tryfoss ........ Merged revisions
368586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-06 01:10 +0000 [r368568] Richard Mudgett
* /, main/features.c: Fix parked call performing a DTMF blind
transfer after being retrieved. When a parked call was retrieved
from the parking lot, it could not do a blind transfer because it
caused the involved calls to be hung up unconditionally. * Made
the ParkedCall application return the ast_bridge_call() return
value. (closes issue ABE-2862) Reported by: Vlad Povorozniuc
........ Merged revisions 368567 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-05 15:27 +0000 [r368524-368536] Kinsey Moore
* /, apps/app_minivm.c: Resolve some build warnings My newly
upgraded compiler caught these usages of uninitialized values.
They weren't actually used. ........ Merged revisions 368533 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* apps/app_voicemail.c, /: Ensure that pages and emails are sent
using RFC822-compliant date format When localization was added to
app_voicemail, these headers were altered when they should have
remained in en_US format for RFC compliance. This reverts the
changes to those two lines. (closes issue ASTERISK-19876)
........ Merged revisions 368520 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-04 22:02 +0000 [r368499] Mark Michelson
* /, channels/chan_sip.c: Relay proper SIP responses on calling
side. Revision 351130 broke corect HANGUPCAUSE setting for the
404 case in chan_sip. Other cases were also potentially broken.
This patch fixes the relaying of causes to be what they used to
be. (closes issue ASTERISK-19914) Reported by Pavel Troller
Tested by Walter Doekes (via a reviewboard test to be committed
later) Patches: chan_sip.diff uploaded by Pavel Troller (license
#6302) ........ Merged revisions 368498 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-04 21:11 +0000 [r368407-368470] Richard Mudgett
* /, UPGRADE.txt: Document BLINDTRANSFER behavior change. (issue
ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call
........ Merged revisions 368469 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* main/channel.c, /: Fix potential deadlock between masquerade and
chan_local. * Restructure ast_do_masquerade() to not hold channel
locks while it calls ast_indicate(). * Simplify many calls to
ast_do_masquerade() since it will never return a failure now. If
it does fail internally because a channel driver callback
operation failed, the only thing ast_do_masquerade() can do is
generate a warning message about strange things may happen and
press on. * Fixed the call to ast_bridged_channel() in
ast_do_masquerade(). This change fixes half of the deadlock
reported in ASTERISK-19801 between masquerades and chan_iax.
(closes issue ASTERISK-19537) Reported by: rmudgett Tested by:
rmudgett Review: https://reviewboard.asterisk.org/r/1915/
........ Merged revisions 368405 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-01 23:24 +0000 [r368310] Richard Mudgett
* /, apps/app_stack.c: Fix deadlock when Gosub used with alternate
dialplan switches. Attempting to remove a channel from
autoservice with the channel lock held will result in deadlock. *
Restructured gosub_exec() to not call ast_parseable_goto() and
ast_exists_extension() with the channel lock held. (closes issue
ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett
........ Merged revisions 368308 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-01 20:22 +0000 [r368267] Kevin P. Fleming
* /, channels/chan_sip.c: Improve SDP parsing warning messages *
'Unsupported media type' is only reported when that is in fact
the case, not when a supported media type is included in an 'm'
line that has an invalid format. * All warning messages related
to parsing 'm' lines now include the 'm' line contents. * (minor
bugfix) newline added to port-number-zero warning messages. *
Warning messages improved to use RFC-specified terminology for
various items. * Warnings for offers that include more than one
port for a single media type now include the media type. Review:
https://reviewboard.asterisk.org/r/1811/ ........ Merged
revisions 368218 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-01 03:28 +0000 [r368093] Michael L. Young
* /, funcs/func_channel.c: Add documentation to function CHANNEL
for options echocan_mode and buffers The ability to set
"echocan_mode" and "buffers" through the dialplan was added to
chan_dahdi some time ago. This patch adds some documentation to
func_channel. (Closes issue ASTERISK-19911) Reported by: Dale
Noll Tested by: Michael L. Young Patches:
asterisk-19911-branch18.diff uploaded by Michael L. Young
(license 5026) Review: https://reviewboard.asterisk.org/r/1949/
........ Merged revisions 368092 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-31 18:20 +0000 [r368042] Richard Mudgett
* res/ael/pval.c, main/tcptls.c, main/manager.c,
res/res_config_odbc.c, /, channels/chan_sip.c,
channels/chan_agent.c, funcs/func_math.c, main/features.c,
apps/app_queue.c, channels/chan_iax2.c, pbx/pbx_config.c:
Coverity Report: Fix issues for error type REVERSE_INULL (core
modules) * Fixes findings: 0-2,5,7-15,24-26,28-31 (issue
ASTERISK-19648) Reported by: Matt Jordan ........ Merged
revisions 368039 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-30 18:07 +0000 [r367907-367981] Richard Mudgett
* /, channels/sig_pri.c, channels/sig_ss7.c: Use the
DEADLOCK_AVOIDANCE() macro instead. (issue ASTERISK-19854)
........ Merged revisions 367980 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/sig_pri.c, channels/sig_ss7.c: Fix deadlock when
executing CLI "pri show channels" and "ss7 show channels"
commands. * Fix sig_pri_lock_owner() to avoid deadlock properly.
* Code pri_grab() better. * Fix sig_ss7_lock_owner() to avoid
deadlock properly. * Code ss7_grab() better. (closes issue
ASTERISK-19854) Reported by: Jaxon Patches:
jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded
by rmudgett (Modified to do the same thing to sig_ss7) Tested by:
Jaxon ........ Merged revisions 367976 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, apps/app_meetme.c: Coverity Report: Fix issues for error type
REVERSE_INULL (deprecated modules) * Fix only issue pointed out
by deprecated_REVERSE_INULL.txt for app_meetme.c in find_user().
* Change use of %i to %d in sscanf() in find_user(). The use of
%i gives unexpected parsing because it can accept hex, octal, and
decimal integer formats. * Changed other uses of %i in
app_meetme() to use %d for consistency. (issue ASTERISK-19648)
Reported by: Matt Jordan ........ Merged revisions 367906 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-29 18:33 +0000 [r367844] Matthew Jordan
* channels/chan_skinny.c: AST-2012-008: Fix remote crash
vulnerability in chan_skinny When a skinny session is
unregistered, the corresponding device pointer is set to NULL in
the channel private data. If the client was not in the on-hook
state at the time the connection was closed, the device pointer
can later be dereferened if a message or channel event attempts
to use a line's pointer to said device. The patches prevent this
from occurring by checking the line's pointer in message handlers
and channel callbacks that can fire after an unregistration
attempt. (closes issue ASTERISK-19905) Reported by: Christoph
Hebeisen Tested by: mjordan, Damien Wedhorn Patches:
AST-2012-008-1.8.diff uploaded by mjordan (license 6283)
AST-2012-008-10.diff uploaded by mjordan (licesen 6283)
2012-05-25 16:30 +0000 [r367782] Richard Mudgett
* /, channels/chan_iax2.c: AST-2012-007: Fix IAX receiving HOLD
without suggested MOH class crash. * Made schedule_delivery() set
the received frame f->data.ptr to NULL if the datalen is zero. *
Fix queue_signalling() memcpy() size error. * Made
queue_signalling() not use C++ keyword variable names. (closes
issue ASTERISK-19597) Reported by: mgrobecker Patches:
jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: rmudgett, Michael L. Young ........ Merged
revisions 367781 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-25 02:29 +0000 [r367731] Michael L. Young
* /, channels/chan_sip.c: Fix pvt_sip for inbound call to use
peer's allowtransfer setting The pvt_sip allowtransfer was not
being set to that of the peer's setting. Therefore, the global
allowtransfer setting was being used instead which would lead to
calls not being transfered if the global setting was set to 'no'
despite the setting on the peer being 'yes' and vice versa, calls
would be allowed to transfer even if the peer's setting was 'no'
but the global setting was 'yes'. (Closes issue ASTERISK-19856)
Reported by: Jacek Tested by: Michael L. Young, Jacek Patches:
issue-asterisk-19856-branch10-v3.diff uploaded by Michael L.
Young (license 5026) Review:
https://reviewboard.asterisk.org/r/1923/ ........ Merged
revisions 367730 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-24 22:29 +0000 [r367679] Richard Mudgett
* apps/app_dial.c, /, apps/app_queue.c: Fix Dial I option ignored
if dial forked and one fork redirects. The Dial and Queue I
option is intended to block connected line updates and
redirecting updates. However, it is a feature that when a call is
locally redirected, the I option is disabled if the redirected
call runs as a local channel so the administrator can have an
opportunity to setup new connected line information.
Unfortunately, the Dial and Queue I option is disabled for *all*
forked calls if one of those calls is redirected. * Make the Dial
and Queue I option apply to each outgoing call leg independently.
Now if one outgoing call leg is locally redirected, the other
outgoing calls are not affected. * Made Dial not pass any
redirecting updates when forking calls. Redirecting updates do
not make sense for this scenario. * Made Queue not pass any
redirecting updates when using the ringall strategy. Redirecting
updates do not make sense for this scenario. * Fixed deadlock
potential with chan_local when Dial and Queue send redirecting
updates for a local redirect. * Converted the Queue stillgoing
flag to a boolean bitfield. (closes issue ASTERISK-19511)
Reported by: rmudgett Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/1920/ ........ Merged
revisions 367678 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-24 13:32 +0000 [r367562] Matthew Jordan
* apps/app_confbridge.c: Fix crash in ConfBridge when user
announcement is played for more than 2 users A patch introduced
in r354938 made it so that ConfBridge would not attempt to play
sound files if those files did not exist. Unfortunately,
ConfBridge uses the same underlying function, play_sound_helper,
to playback both sound files and numbers to callers. When a
number is being played back, the name of the sound file is
expected to be NULL. This NULL value was passed into a function
that tested for the existance of a sound file and is not tolerant
to NULL file names, causing a crash. This patch fixes the
behavior, such that if a sound file does not exist we do not
attempt to play it, but we only attempt that check if the a sound
file was specified in the first place. If a sound file was not
specified, we use the 'play number' logic in the helper function.
(closes issue ASTERISK-19899) Reported by: Florian Gilcher Tested
by: Florian Gilcher patches: asterisk-19899.diff uploaded by
mjordan (license 6283)
2012-05-23 23:16 +0000 [r367470] Richard Mudgett
* main/pbx.c, /: Fix WaitExten(x,m(musicclass)) string termination.
The AST_CONTROL_HOLD MOH class from the WaitExten application can
now be queued onto a channel, passed over local channels with the
/m option, and passed over IAX channels. ........ Merged
revisions 367469 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-23 20:29 +0000 [r367417] Mark Michelson
* main/tcptls.c, /: Only call SSL_CTX_free if DO_SSL is defined.
Thanks to Paul Belanger for pointing out this error. ........
Merged revisions 367416 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-23 13:25 +0000 [r367369] Matthew Jordan
* /, channels/chan_sip.c, CHANGES, channels/sip/include/sip.h:
Re-add LastMsgsSent value for SIP peers Previously, MWI logic
utilized a counter called 'lastmsgssent' to know whether or not
MWI NOTIFY requests had been sent to a specific peer. When MWI
notifications were changed to use the internal event framework,
this value was no longer needed for its original purpose. Hence,
it was no longer updated with the new/old message counts for a
peer. The value was previously removed for Asterisk 10; however,
since it was still present in Asterisk 1.8 and still useful for
reporting purposes, it was decided to re-add the value. This
patch re-adds the 'LastMsgsSent' field in the response to an
AMI/CLI 'sip show peer [peer]' command, and makes it so that the
value of lastmsgssent is updated appropriately. The value should
now display the new/old message counts for a particular peer.
(closes issue ASTERISK-17866) Reported by: Steve Davies patches
by: ast-17866-rb1272.patch (License #5041 by irroot) Modified
slightly for this commit Review:
https://reviewboard.asterisk.org/r/1939 ........ Merged revisions
367362 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-22 17:21 +0000 [r367267-367299] Terry Wilson
* main/channel.c, /, include/asterisk/cel.h,
include/asterisk/channel.h, main/cel.c, main/asterisk.c: Fix race
condition for CEL LINKEDID_END event This patch fixes to
situations that could cause the CEL LINKEDID_END event to be
missed. 1) During a core stop gracefully, modules are unloaded
when ast_active_channels == 0. The LINKDEDID_END event fires
during the channel destructor. This means that occasionally, the
cel_* module will be unloaded before the channel is destroyed. It
seemed generally useful to wait until the refcount of all
channels == 0 before unloading, so I added a channel counter and
used it in the shutdown code. 2) During a masquerade,
ast_channel_change_linkedid is called. It calls
ast_cel_check_retire_linkedid which unrefs the linkedid in the
linkedids container in cel.c. It didn't ref the new linkedid. Now
it does. Review: https://reviewboard.asterisk.org/r/1900/
........ Merged revisions 367292 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_sip.c: Resolve crash in subscribing for MWI
notifications ASTOBJ_UNREF sets the variable to NULL after
unreffing it, so the variable should definitely not be used after
that. To solve this in the two cases that affect subscribing for
MWI notifications, we instead save the ref locally, and unref
them in the error conditions. (closes issue ASTERISK-19827)
Reported by: B. R Review:
https://reviewboard.asterisk.org/r/1940/ ........ Merged
revisions 367266 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-18 17:50 +0000 [r367003-367028] Mark Michelson
* channels/chan_dahdi.c, /, main/say.c: Address MISSING_BREAK
static analysis reports some more. This addresses core findings 4
and 6. Moises Silva helped me by stating that a break could be
safely added to the case where it is added in chan_dahdi.c In
say.c, I have added a comment indicating that static analysis
complains but that it is currently unknown if this is correct.
This fixes all core findings of this type. (closes issue
ASTERISK-19662) reported by Matthew Jordan ........ Merged
revisions 367027 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h:
Fix memory leak of SSL_CTX structures in TLS core. SSL_CTX
structures were allocated but never freed. This was a bigger
issue for clients than servers since new SSL_CTX structures could
be allocated for each connection. Servers, on the other hand,
typically set up a single SSL_CTX for their lifetime. This is
solved in two ways: 1. In __ssl_setup(), if a tcptls_cfg has an
ssl_ctx on it, it is freed so that a new one can take its place.
2. A companion to ast_ssl_setup() called ast_ssl_teardown() has
been added so that servers can properly free their SSL_CTXs.
(issue ASTERISK-19278) ........ Merged revisions 367002 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-18 15:45 +0000 [r366948] Matthew Jordan
* main/cli.c, /, channels/chan_sip.c, funcs/func_odbc.c: Fix more
memory leaks This patch adds to what was fixed in r366880.
Specifically, it addresses the following: * chan_sip: dispose of
an allocated frame in off nominal code paths in sip_rtp_read *
func_odbc: when disposing of an allocated resultset, ensure that
any rows that were appended to that resultset are also disposed
of * cli: free the created return string buffer in another off
nominal code path (issue ASTERISK-19665) Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1922/ ........ Merged
revisions 366944 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-18 14:18 +0000 [r366884] Kinsey Moore
* /, channels/sip/config_parser.c: Reorder and renumber tests
appropriately It appears that a patch did not apply properly when
adding tests 12 and 13 and test 11 was duplicated. These tests
have been reordered and renumbered such that they make sense.
........ Merged revisions 366882 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-18 14:01 +0000 [r366881] Matthew Jordan
* main/xmldoc.c, apps/app_voicemail.c, res/res_calendar.c,
main/netsock2.c, res/res_rtp_asterisk.c, main/pbx.c,
res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
apps/app_page.c, /, funcs/func_dialgroup.c, channels/chan_sip.c,
apps/app_record.c, res/res_calendar_caldav.c,
res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c,
res/res_jabber.c, main/editline/term.c, main/enum.c,
main/config.c, res/res_srtp.c, main/cli.c,
main/editline/tokenizer.c, main/data.c, channels/chan_dahdi.c,
funcs/func_odbc.c, main/features.c, apps/app_minivm.c,
main/editline/readline.c, channels/sip/config_parser.c: Fix a
variety of memory leaks This patch addresses a number of memory
leaks in a variety of modules that were found by a static
analysis tool. A brief summary of the changes: * app_minivm: free
ast_str objects on off nominal paths * app_page: free the
ast_dial object if the requested channel technology cannot be
appended to the dialing structure * app_queue: if a penalty rule
failed to match any existing rule list names, the created rule
would not be inserted and its memory would be leaked * app_read:
dispose of the created silence detector in the presence of off
nominal circumstances * app_voicemail: dispose of an allocated
unique ID field for MWI event un-subscribe requests in off
nominal paths; dispose of configuration objects when using the
secret.conf option * chan_dahdi: dispose of the allocated frame
produced by ast_dsp_process * chan_iax2: properly unref peer in
CLI command "iax2 unregister" * chan_sip: dispose of the
allocated frame produced by sip_rtp_read's call of
ast_dsp_process; free memory in parse unit tests *
func_dialgroup: properly deref ao2 object grhead in nominal path
of dialgroup_read * func_odbc: free resultset in off nominal
paths of odbc_read * cli: free match_list in off nominal paths of
CLI match completion * config: free comment_buffer/list_buffer
when configuration file load is unchanged; free the same buffers
any time they were created and config files were processed *
data: free XML nodes in various places * enum: free context
buffer in off nominal paths * features: free ast_call_feature in
off nominal paths of applicationmap config processing * netsock2:
users of ast_sockaddr_resolve pass in an ast_sockaddr struct that
is allocated by the method. Failures in ast_sockaddr_resolve
could result in the users of the method not knowing whether or
not the buffer was allocated. The method will now not allocate
the ast_sockaddr struct if it will return failure. * pbx: cleanup
hash table traversals in off nominal paths; free ignore pattern
buffer if it already exists for the specified context * xmldoc:
cleanup various nodes when we no longer need them *
main/editline: various cleanup of pointers not being freed before
being assigned to other memory, cleanup along off nominal paths *
menuselect/mxml: cleanup of value buffer for an attribute when
that attribute did not specify a value * res_calendar*: responses
are allocated via the various *_request method returns and should
not be allocated in the various write_event methods; ensure
attendee buffer is freed if no data exists in the parsed node;
ensure that calendar objects are de-ref'd appropriately *
res_jabber: free buffer in off nominal path * res_musiconhold:
close the DIR* object in off nominal paths * res_rtp_asterisk: if
we run out of ports, close the rtp socket object and free the rtp
object * res_srtp: if we fail to create the session in libsrtp,
destroy the temporary ast_srtp object (issue ASTERISK-19665)
Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1922 ........ Merged revisions
366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-17 14:41 +0000 [r366792] Jonathan Rose
* /, channels/chan_sip.c: chan_sip: Fix missed locking of opposing
pvt for directmedia acl from r366547 It also required deadlock
avoidance since two sip_pvts structs needed to be locked
simultaneously. Trunk handles it differently, so this is a 1.8
and 10 patch only. ........ (issue AST-876) Merged revisions
366791 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-17 12:57 +0000 [r366741] Matthew Jordan
* channels/chan_dahdi.c, /, res/res_calendar_ews.c: Fix checking
bounds of array index after using it; improper sizeof This patch
fixes two problems pointed out by a static analysis tool. * In
chan_dahdi, when an event is handled the index of the sub channel
is first obtained. In very off nominal cases, the method that
determines the index can return a negative value. In the event
handling code, whether or not the index returned is valid was
being checked after that value was used to index into an array.
This patch makes it so the value is checked before any indexing
is done. * In res_calendar_ews, sizeof was being passed a pointer
instead of the struct to determine the amount of memory to
allocate. (issue ASTERISK-19651) Reported by: Matt Jordan (closes
issue ASTERISK-19671) Reported by: Matt Jordan ........ Merged
revisions 366740 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-15 23:39 +0000 [r366598] Mark Michelson
* /, channels/chan_sip.c: Correct misuse of ast_strip_quoted() when
getting a Diversion header's reason parameter. The use here was
assuming that the pointer would be updated, but the updated
string is actually returned by ast_strip_quoted() instead.
........ Merged revisions 366597 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-15 20:44 +0000 [r366591] Jonathan Rose
* /, channels/chan_sip.c: chan_sip: Check the right channel's host
address for directmediapermit/deny Prior to this patch, when
checking the addresses for directmediapermit and
denydirectmediadeny, Asterisk would check the host address of the
channel permit/deny was specified, which defers from the
expectations of both our users and the development team. Instead,
directmediapermit/deny now checks against the address of the
channel that the peer with the ACL is connected to. (issue
AST-876) Review: https://reviewboard.asterisk.org/r/1899/
........ Merged revisions 366547 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-14 20:06 +0000 [r366390-366412] Mark Michelson
* /, pbx/dundi-parser.c: Fix two more coverity constant expression
result findings. These correspond to findings 0 and 1 in the core
findings of ASTERISK-19649. After contacting Mark Spencer, he was
unsure of what the intent behind these lines of code were, so
they are being axed. For Asterisk 1.8 and 10, the output of
debugging DUNDi frames will not be changed, but for trunk the
"Retry" portion will be omitted since it does not properly
distinguish retransmissions from initial frames. (closes issue
ASTERISK-19649) Reported by Matthew Jordan ........ Merged
revisions 366409 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_sip.c: Fix broken reinvite glare scenario. To
make a long story short, reinvite glares were broken because
Asterisk would invert the To and From headers when ACKing a 491
response. The reason was because the initreq of the dialog was
being changed to the incoming glared reinvite instead of being
set to the outgoing glared reinvite. This change has three parts
* In handle_incoming, we never will reject an ACK because it has
a to-tag present, even if we think the request may be out of
dialog. * In handle_request_invite, we do not change the initreq
when receiving a reinvite to which we will respond with a 491. *
In handle_request_invite, several superflous settings up
pendinginvite have been removed since this is dones automatically
by transmit_response_reliable Review:
https://reviewboard.asterisk.org/r/1911 ........ Merged revisions
366389 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-11 23:59 +0000 [r366297] Russell Bryant
* /, addons/format_mp3.c: format_mp3: Fix a possible crash
mp3_read(). This patch fixes a potential crash in mp3_read() by
not assuming that dbuf has enough data to finish filling up the
output buffer. The patch also makes sure that the dbuf state gets
reset after we know we read everything out of it already. In
passing, this patch includes some other cleanups of this module,
including stripping trailing whitespace, formatting fixes based
on coding guidelines, and removing a number of unused members
from the private state struct. (closes issue ASTERISK-19761)
Reported by: Chris Maciejewsk Tested by: Chris Maciejewsk
........ Merged revisions 366296 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-10 23:42 +0000 [r366241] Richard Mudgett
* main/channel.c, /: * Made ast_change_name() hold the channels
container lock while changing the channel name. * Eliminate
redundant list not empty check in clone_variables(). ........
Merged revisions 366240 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-10 20:54 +0000 [r366168] Kinsey Moore
* main/xmldoc.c, apps/app_voicemail.c, funcs/func_speex.c,
main/pbx.c, res/res_calendar_icalendar.c, /, channels/chan_sip.c,
funcs/func_lock.c, channels/chan_agent.c,
channels/sip/reqresp_parser.c, main/devicestate.c,
pbx/dundi-parser.c, channels/chan_iax2.c, channels/iax2-parser.c,
main/config.c, res/res_monitor.c, main/cdr.c, main/channel.c,
res/ael/pval.c, main/data.c, channels/chan_dahdi.c,
main/tcptls.c, main/manager.c, main/features.c, main/app.c,
main/event.c, pbx/pbx_dundi.c, res/res_odbc.c: Resolve
FORWARD_NULL static analysis warnings This resolves core findings
from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28,
30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111,
and 115. Finding numbers 26, 33, and 29 were already resolved.
Those skipped were either extended/deprecated or in areas of code
that shouldn't be disturbed. (Closes issue ASTERISK-19650)
........ Merged revisions 366167 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-10 16:55 +0000 [r366106] Jonathan Rose
* main/xmldoc.c, apps/app_voicemail.c, main/pbx.c,
channels/sig_analog.c, /, channels/chan_sip.c, funcs/func_lock.c,
main/features.c, main/acl.c, channels/iax2-provision.c,
apps/app_queue.c, channels/chan_iax2.c, res/ael/ael.flex,
funcs/func_devstate.c, main/asterisk.c: Coverity Report: Fix
issues for error type CHECKED_RETURN for core (issue
ASTERISK-19658) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1905/ ........ Merged
revisions 366094 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-10 16:13 +0000 [r366053] Mark Michelson
* /, channels/chan_sip.c: Close the proper tcptls_session when
session creation fails. (issue AST-998) Reported by: Thomas
Arimont Tested by: Thomas Arimont ........ Merged revisions
366052 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-10 15:43 +0000 [r365990-366049] Jonathan Rose
* /, apps/app_page.c, funcs/func_cdr.c, main/features.c,
apps/app_disa.c, apps/app_chanspy.c: Coverity Report: Fix issues
for error type UNINIT in Core supported modules (issue
ASTERISK-19652) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1909/ ........ Merged
revisions 366048 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, codecs/codec_dahdi.c: Block on frameout if the hardware has
enough samples to complete a frame. Fixes some problems with
skipping audio in elaborate scenarios involving multiple codecs
by making codec_dahdi operate in a more synchronous fashion
similar to codec_g729. This change also fixes the use of file
conversion tools from Asterisk's CLI. This change may cause the
thread responsible for transcoding audio to block briefly (Shaun
Ruffell describes this as 'several milliseconds') while waiting
for the hardware transcoder. (closes issue ASTERISK-19643)
reported by: Shaun Ruffell Patches:
0001-codec_dahdi-Block-on-frameout-the-hardware-has-enoug.patch
uploaded by Shaun Ruffell (license 5417) ........ Merged
revisions 365989 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-09 16:15 +0000 [r365898] Mark Michelson
* /, channels/chan_sip.c: Prevent sip_pvt refleak when an
ast_channel outlasts its corresponding sip_pvt. chan_sip was
coded under the assumption that a SIP dialog with an owner
channel will always be destroyed after the owner channel has been
hung up. However, there are situations where the SIP dialog can
time out and auto destruct before the corresponding channel has
hung up. A typical example of this would be if the 'h' extension
in the dialplan takes a long time to complete. In such cases,
__sip_autodestruct() would complain about the dialog being auto
destroyed with an owner channel still in place. The problem is
that even once the owner channel was hung up, the sip_pvt would
still be linked in its ao2_container because nothing would ever
unlink it. The fix for this is that if __sip_autodestruct() is
called for a sip_pvt that still has an owner channel in place,
the destruction is rescheduled for 10 seconds in the future. This
will continue until the owner channel is finally hung up. (closes
issue ASTERISK-19425) reported by David Cunningham Patches:
ASTERISK-19425.patch uploaded by Mark Michelson (License #5049)
(closes issue ASTERISK-19455) reported by Dean Vesvuio Tested by
Dean Vesvuio ........ Merged revisions 365896 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-08 20:25 +0000 [r365632-365701] Richard Mudgett
* /, apps/app_followme.c: * Fix FollowMe memory leak on error paths
in app_exec(). * Fix FollowMe leaving recorded caller name file
on error paths in app_exec(). * Use correct buffer dimension
define in struct call_followme.moh[] and struct
fm_args.namerecloc[]. This fixes unexpected namerecloc filename
length restriction. ........ Merged revisions 365692 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, apps/app_followme.c: * Fix accept/decline DTMF buffer
overwrite in FollowMe. * Made use MAX_YN_STRING define to make
all accept/decline DTMF buffers the same size. Just using 20
isn't good enough when someone didn't get the memo. * Fix stupid
use of a global variable in FollowMe. (ynlongest) * Fix bit field
declarations in FollowMe. ........ Merged revisions 365631 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-08 15:51 +0000 [r365575] Mark Michelson
* /, channels/chan_sip.c: Send more accurate identification
information in dialog-info SIP NOTIFYs. This uses the calling
channel's caller ID and connected line information to populate
the remote and local identities in the dialog-info NOTIFY when an
extension is ringing. There is a bit of an oddity here, and that
is that we seed the remote target with the To header of the
outbound call rather than the from header. This is because it was
reported that seeding with the from header caused hints to be
broken with certain SNOM devices. A comment has been added to the
code to explain this. (closes issue ASTERISK-16735) reported by
Maciej Krajewski patches: local_remote_hint2.diff uploaded by
Mark Michelson (license #5049) 16735_tweak1.diff uploaded by Mark
Michelson (license #5049) Tested by Niccolo Belli ........ Merged
revisions 365574 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-07 18:43 +0000 [r365478] Richard Mudgett
* /, tests/test_config.c: Fix type punned compiler warning in
test_config.c ........ Merged revisions 365476 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-07 18:39 +0000 [r365475] Matthew Jordan
* apps/app_voicemail.c, main/pbx.c, /: Support VoiceMail d() option
when extension does not exist in channel's context The VoiceMail
d([c]) option is documented to accept digits for a new extension
in context , if played during the greeting. This option works
fine if the extension being redirected to has an extension with
the same initial digit in the channel's current context. If that
digit did not happen to exist in some extension, a dialplan match
would fail and the user would not be redirected. This patch fixes
it such that if the option is used, the extensions are
matched in that context as opposed to the caller's original
context. (closes issue ASTERISK-18243) Reported by: mjordan
Tested by: mjordan Review:
https://reviewboard.asterisk.org/r/1892 ........ Merged revisions
365474 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-04 22:15 +0000 [r365399] Kinsey Moore
* apps/app_voicemail.c, /, channels/chan_sip.c, funcs/func_aes.c,
main/features.c, apps/app_followme.c, channels/chan_iax2.c,
channels/sip/config_parser.c, pbx/pbx_config.c,
apps/app_chanspy.c, apps/app_stack.c, main/config.c: Fix many
issues from the NULL_RETURNS Coverity report Most of the changes
here are trivial NULL checks. There are a couple optimizations to
remove the need to check for NULL and outboundproxy parsing in
chan_sip.c was rewritten to avoid use of strtok. Additionally, a
bug was found and fixed with the parsing of outboundproxy when
"outboundproxy=," was set. (Closes issue ASTERISK-19654) ........
Merged revisions 365398 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-04 16:28 +0000 [r365320] Richard Mudgett
* channels/chan_local.c, /: Fix local channel chains optimizing
themselves out of a call. * Made chan_local.c:check_bridge()
check the return value of ast_channel_masquerade(). In long
chains of local channels, the masquerade occasionally fails to
get setup because there is another masquerade already setup on an
adjacent local channel in the chain. * Made the outgoing local
channel (the ;2 channel) flush one voice or video frame per
optimization attempt. * Made sure that the outgoing local channel
also does not have any frames in its queue before the masquerade.
* Made do the masquerade immediately to minimize the chance that
the outgoing channel queue does not get any new frames added and
thus unconditionally flushed. * Made block indication -1 (Stop
tones) event when the local channel is going to optimize itself
out. When the call is answered, a chain of local channels pass
down a -1 indication for each bridge. This blizzard of -1 events
really slows down the optimization process. (closes issue
ASTERISK-16711) Reported by: Alec Davis Tested by: rmudgett, Alec
Davis Review: https://reviewboard.asterisk.org/r/1894/ ........
Merged revisions 365313 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-04 15:51 +0000 [r365299] Mark Michelson
* res/res_rtp_asterisk.c, /: Fix core FINDING 2, FINDING 3, and
FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESULT report.
These three all are in RTP code that attempts to print the number
of sequence number cycles in an RTCP RR report. The code was
masking out the upper 16 bits and then shifting the number right
by 16 bits. This led to an all zero result in all cases. The fix
is to do the shift without the bit masking. (issue
ASTERISK-19649) ........ Merged revisions 365298 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-03 15:01 +0000 [r365155-365160] Alexandr Anikin
* addons/ooh323c/src/ooh323.c, /,
addons/ooh323c/src/h323/H323-MESSAGES.h,
addons/ooh323c/src/h323/H323-MESSAGESEnc.c: Fix warning of
Coverity Static analysis, change H225ProtocolIdentifier from
value to pointer per functions that use this. (close issue
ASTERISK-19670) Reported by: Matt Jordan Patches:
ASTERISK-19670.patch (License #5415) ........ Merged revisions
365159 from http://svn.asterisk.org/svn/asterisk/branches/1.8
* addons/ooh323c/src/ooq931.c, /: Fix coverity static analysis
warning, allocate full ie structure instead of without data
buffer (close issue ASTERISK-19674) Reported by: Matt Jordan
Patches: ASTERISK-19674.patch (License #5415) ........ Merged
revisions 365143 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-04 Asterisk Development Team
* Asterisk 10.5.0 Released.
2012-05-30 Asterisk Development Team
* Asterisk 10.5.0-rc2 Released.
* Resolve crash in subscribing for MWI notifications.
ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the
variable shoudl definitely not be used after that. To solve this in
the two cases that affect subscribing for MWI notifications, we
instead save the ref locally, and unref them in the error
conditions.
(closes issue ASTERISK-19827)
Reported by: B. R.
Review: https://reviewboard.asterisk.org/r/1940/
* Fix crash in ConfBridge when user announcement is played for more
than 2 users
A patch introduced in r354938 made it so that ConfBridge would not
attempt to play sound files if those files did not exist.
Unfortunately, ConfBridge uses the same underlying fucntion,
play_sound_helper, to playback both the sound files and numbers to
callers. When a number is being played back, the name of the sound
file is expected to be NULL. This NULL value was passed into a
function that tested for the existance of a sound file and is not
tolerant to NULL file names, causing a crash.
This patch fixes the behavior, such that if a sound file does not
exist we do not attempt to play it, but we only attempt that check
if the sound file was specified in the first place. If a sound file
was not specified, we use the 'play number' logic in the helper
function.
(closes issue ASTERISK-19899)
Reported by: Florian Gilcher
Tested by: Florian Gilcher
patches:
ASTERISK-19899.diff uploaded by mjordan (license 6283)
* AST-2012-007
* AST-2012-008
2012-05-03 Asterisk Development Team
* Asterisk 10.5.0-rc1 Released.
2012-05-02 17:29 +0000 [r365083] Terry Wilson
* channels/chan_local.c, /, main/cel.c: Multiple revisions
365006,365068 ........ r365006 | twilson | 2012-05-02 10:49:03
-0500 (Wed, 02 May 2012) | 12 lines Fix a CEL LINKEDID_END race
and local channel linkedids This patch has the ;2 channel inherit
the linkedid of the ;1 channel and fixes the race condition by no
longer scanning the channel list for "other" channels with the
same linkedid. Instead, cel.c has an ao2 container of linkedid
strings and uses the refcount of the string as a counter of how
many channels with the linkedid exist. Not only does this
eliminate the race condition, but it also allows us to look up
the linkedid by the hashed key instead of traversing the entire
channel list. Review: https://reviewboard.asterisk.org/r/1895/
........ r365068 | twilson | 2012-05-02 12:02:39 -0500 (Wed, 02
May 2012) | 11 lines Don't leak a ref if out of memory and can't
link the linkedid If the ao2_link fails, we are most likely out
of memory and bad things are going to happen. Before those bad
things happen, make sure to clean up the linkedid references.
This patch also adds a comment explaining why linkedid can't be
passed to both local channel allocations and combines two ao2_ref
calls into 1. Review: https://reviewboard.asterisk.org/r/1895/
........ Merged revisions 365006,365068 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-02 16:16 +0000 [r365014] Michael L. Young
* tests/test_security_events.c: Update security events unit tests
The security events framework API was changed in Asterisk 10 but
the unit tests were not updated at the same time. This patch does
the following: * Adds two more security events that were added to
the API * Add challenge, received_challenge and received_hash in
the inval_password security event unit test (issue
ASTERISK-19760) Reported by: Michael L. Young Tested by: Michael
L. Young Patches: issue-asterisk-19760-branch10.diff uploaded by
Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/1877/
2012-05-02 02:44 +0000 [r364965] Matthew Jordan
* main/audiohook.c: Only log a failure to get read/write samples
from factories if it didn't happen In audiohook_read_frame_both,
anytime samples are obtained from the read/write factories a
debug statement is logged stating that samples were not obtained
from the factories. This statement used to only occur if
option_debug was turned on and no samples were obtained; in some
refactoring when the option_debug statement was removed, the
"else" clause was removed as well. This patch makes it so that
those debug log statements only occur if the condition leading up
to them actually happened.
2012-05-01 23:14 +0000 [r364903] Richard Mudgett
* /, main/astobj2.c: Fixed __ao2_ref() validating user_data twice.
(closes issue ASTERISK-19755) Reported by: Gunther Kelleter
Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther
Kelleter ........ Merged revisions 364902 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-01 23:10 +0000 [r364900] Mark Michelson
* /, funcs/func_volume.c: Fix Coverity-reported ARRAY_VS_SINGLETON
error. As it turned out, this wasn't a huge deal. We were calling
ast_app_parse_options() for a set of options of which none took
arguments. The proper thing to do for this case is to pass NULL
for the "args" parameter here. We were instead passing a
seemingly-randomly chosen char * from the function. While this
would never get written to, you can rest assured things would
have gotten bad had new options (which took arguments) been added
to func_volume. (closes issue ASTERISK-19656) ........ Merged
revisions 364899 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-01 21:50 +0000 [r364845] Richard Mudgett
* channels/chan_local.c, /: * Fix error path resouce leak in
local_request(). * Restructure local_request() to reduce
indentation. ........ Merged revisions 364840 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-01 21:44 +0000 [r364842] Jason Parker
* main/manager.c, /: Prevent a potential crash when using manager
hooks. Found by me while poking at DPMA-127. ........ Merged
revisions 364841 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-01 19:07 +0000 [r364787] Kinsey Moore
* /, apps/app_confbridge.c: Play conf-placeintoconf message to the
correct channel Correct the code in app_confbridge to play the
conf-placeintoconf message to the marked user entering the bridge
instead of to the conference while the marked user hears silence.
(closes issue ASTERISK-19641) Reported-by: Mark A Walters
........ Merged revisions 364786 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-01 18:23 +0000 [r364777] Jonathan Rose
* /, main/app.c: Fix bad check in voicemail functions for
ast_inboxcount2_func Check looks for ast_inboxcount_func instead
of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes
issue ASTERISK-19718) Reported by: Corey Farrell Patches:
ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell
(license 5909) ........ Merged revisions 364769 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-30 19:42 +0000 [r364707] Mark Michelson
* /, channels/chan_sip.c: Revert improved identities sent in
dialog-info NOTIFY requests in r360862 Revision 360862 was
intended to improve identities sent in dialog-info NOTIFY
requests. Some users reported that hint became broken once this
was done. It's not clear exactly what part of the patch has
caused this regression, but broken hints are bad. For now, this
revision is being reverted so that the next releases of Asterisk
do not have bad behavior in them. The original reported issue
will have to be fixed differently in the next version of
Asterisk. (issue ASTERISK-16735) ........ Merged revisions 364706
from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-30 16:48 +0000 [r364651] Alexandr Anikin
* /, addons/ooh323cDriver.c: Fix use freed pointer in return value
from call thread (issue ASTERISK-19663) Reported by: Matt Jordan
Patches: ASTERISK-19663-ooh323.patch (License #5415) ........
Merged revisions 364649 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-30 16:43 +0000 [r364650] Mark Murawki
* /, main/logger.c: Merged revisions 364635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) |
10 lines Sanatize result from bfd_find_nearest_line
(BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file
to null resulting in a crash when strrchr(file) runs (closes
issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark
Murawski ........
2012-04-29 19:43 +0000 [r364579] Matthew Jordan
* formats/format_g719.c, formats/format_siren7.c,
formats/format_g729.c, formats/format_ilbc.c, /,
formats/format_sln.c, formats/format_vox.c, formats/format_wav.c,
formats/format_pcm.c, formats/format_g723.c,
formats/format_h263.c, formats/format_h264.c,
formats/format_wav_gsm.c, formats/format_siren14.c,
formats/format_gsm.c: Fix error that caused truncate operations
to fail Another very inappropriate placement of a ')' (again
introduced in r362151) caused the various truncate operations to
attempt to truncate the sound file at a position of '0'. (issue
ASTERISK-19655) Reported by: Matt Jordan (issue ASTERISK-19810)
Reported by: colbec ........ Merged revisions 364578 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-29 02:21 +0000 [r364536] Michael L. Young
* apps/confbridge/conf_config_parser.c: Fix configuring custom
sound_leader_has_left in confbridge.conf The configuration option
to specify a custom sound_leader_has_left file for a conference
bridge was not being parsed. This patch fixes it so that a custom
sound file will now be used. (closes issue ASTERISK-19771)
Reported by: Pawel Kuzak Tested by: Pawel Kuzak, Michael L. Young
Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak
(license 6380) Review: https://reviewboard.asterisk.org/r/1884/
2012-04-27 22:33 +0000 [r364365-364369] Terry Wilson
* tests/test_config.c (added): Add missing test_config.c
* /, main/config.c: Fix ast_parse_arg numeric type range checking
and add tests ast_parse_arg wasn't checking for strto* parse
errors or limiting the results by the actual range of the numeric
types. This patch fixes that and adds unit tests as well. Review:
https://reviewboard.asterisk.org/r/1879/ ........ Merged
revisions 364340 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-27 21:58 +0000 [r364342] Mark Michelson
* /, channels/chan_sip.c: Don't attempt to make use of the
dynamic_exclude_static ACL if DNS lookup fails. (closes issue
ASTERISK-18321) Reported by Dan Lukes Patches:
ASTERISK-18321.patch by Mark Michelson (license #5049) ........
Merged revisions 364341 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-27 19:30 +0000 [r364285] Matthew Jordan
* include/asterisk/time.h, /: Prevent overflow in calculation in
ast_tvdiff_ms on 32-bit machines The method ast_tvdiff_ms
attempts to calculate the difference, in milliseconds, between
two timeval structs, and return the difference in a 64-bit
integer. Unfortunately, it assumes that the long tv_sec/tv_usec
members in the timeval struct are large enough to hold the
calculated values before it returns. On 64-bit machines, this
might be the case, as a long may be 64-bits. On 32-bit machines,
however, a long may be less (32-bits), in which case, the
calculation can overflow. This overflow caused significant
problems in MixMonitor, which uses the method to determine if an
audio factory, which has not presented audio to an audiohook, is
merely late in providing said audio or will never provide audio.
In an overflow situation, the audiohook would incorrectly
determine that an audio factory that will never provide audio is
merely late instead. This led to situations where a MixMonitor
never recorded any audio. Note that this happened most frequently
when that MixMonitor was started by the ConfBridge application
itself, or when the MixMonitor was attached to a Local channel.
(issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben
Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license
#6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark
Murawski Tested by: Michael L. Young Patches:
32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan)
(closes issue ASTERISK-19471) Reported by: feyfre Tested by:
feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review:
https://reviewboard.asterisk.org/r/1889/ ........ Merged
revisions 364277 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-27 18:58 +0000 [r364259] Kinsey Moore
* /, channels/chan_sip.c: Allow SIP pvts involved in Replaces
transfers to fall out of reference sooner Unref the SIP pvt
stored in the refer structure as soon as it is no longer needed
so that the pvt and associated file descriptors can be freed
sooner. This change makes a reference decrement unnecessary in
code that handles SIP BYE/Also transfers which should not touch
the reference anyway. (related to issue ASTERISK-19579) ........
Merged revisions 364258 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-27 14:44 +0000 [r364204] Matthew Jordan
* /, channels/chan_sip.c: Allow for reloading SRTP crypto keys
within the same SIP dialog As a continuation of the patch in
r356604, which allowed for the reloading of SRTP keys in
re-INVITE transfer scenarios, this patch addresses the more
common case where a new key is requested within the context of a
current SIP dialog. This can occur, for example, when certain
phones request a SIP hold. Previously, once a dialog was
associated with an SRTP object, any subsequent attempt to process
crypto keys in any SDP offer - either the current one or a new
offer in a new SIP request - were ignored. This patch changes
this behavior to only ignore subsequent crypto keys within the
current SDP offer, but allows future SDP offers to change the
keys. (issue ASTERISK-19253) Reported by: Thomas Arimont Tested
by: Thomas Arimont Review:
https://reviewboard.asteriskorg/r/1885/ ........ Merged revisions
364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-27 12:54 +0000 [r364163] Stefan Schmidt
* res/res_calendar_icalendar.c, res/res_calendar_caldav.c: fix a
wrong behavior of alarm timezones in caldav and icalendar when an
alarm doesnt use utc. This change uses the same timezone from the
start time.
2012-04-26 21:10 +0000 [r364065-364109] Richard Mudgett
* /, apps/app_directed_pickup.c: Update Pickup application
documentation. (With feeling this time.) ........ Merged
revisions 364108 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, main/features.c: Fix DTMF atxfer running h exten after the
wrong bridge ends. When party B does an attended transfer of
party A to party C, the attending bridge between party B and C
should not be running an h exten when the bridge ends. Running an
h exten now sets a softhangup flag to ensure that an AGI will run
in dead AGI mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the
party B channel for the attending bridge between party B and C.
(closes issue AST-870) (closes issue ASTERISK-19717) Reported by:
Mario (closes issue ASTERISK-19633) Reported by: Andrey Solovyev
Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch
uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario
........ Merged revisions 364060 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-26 19:30 +0000 [r364047] Terry Wilson
* /, main/asterisk.c: Add more constness to the end_buf pointer in
the netconsole issue ASTERISK-18308 Review:
https://reviewboard.asterisk.org/r/1876/ ........ Merged
revisions 364046 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-26 13:27 +0000 [r363987] Kinsey Moore
* /, channels/chan_sip.c: Fix reference leaks involving SIP
Replaces transfers The reference held for SIP blind transfers
using the Replaces header in an INVITE was never freed on success
and also failed to be freed in some error conditions. This caused
a file descriptor leak since the RTP structures in use at the
time of the transfer were never freed. This reference leak and
another relating to subscriptions in the same code path have now
been corrected. (closes issue ASTERISK-19579) ........ Merged
revisions 363986 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-26 09:46 +0000 [r363935] Alec L Davis
* /, channels/chan_sip.c: chan_sip: [general] maxforwards, not
checked for a value greater than 255 The peer maxforwards is
checked for both '< 1' and '> 255', but the default 'maxforwards'
in the [general] section is only checked for '< 1' alecdavis
(license 585) Reported by: alecdavis Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1888/ ........ Merged
revisions 363934 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-26 03:11 +0000 [r363376-363876] Richard Mudgett
* /, apps/app_directed_pickup.c: Update Pickup application
documentation. (Even better) ........ Merged revisions 363875
from http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, apps/app_directed_pickup.c: Update Pickup application
documentation. ........ Merged revisions 363788 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* channels/chan_dahdi.c, /, channels/sig_pri.c: Make
DAHDISendCallreroutingFacility wait 5 seconds for a reply before
disconnecting the call. Some switches may not handle the
call-deflection/call-rerouting message if the call is
disconnected too soon after being sent. Asteisk was not waiting
for any reply before disconnecting the call. * Added a 5 second
delay before disconnecting the call to wait for a potential
response if the peer does not disconnect first. (closes issue
ASTERISK-19708) Reported by: mehdi Shirazi Patches:
jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: rmudgett ........ Merged revisions 363730
from http://svn.asterisk.org/svn/asterisk/branches/1.8
* channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c:
Clear ISDN channel resetting state if the peer continues to use
it. Some ISDN switches occasionally fail to send a RESTART
ACKNOWLEDGE in response to a RESTART request. * Made the second
SETUP received after sending a RESTART request clear the channel
resetting state as if the peer had sent the expected RESTART
ACKNOWLEDGE before continuing to process the SETUP. The peer may
not be sending the expected RESTART ACKNOWLEDGE. (issue
ASTERISK-19608) (issue AST-844) (issue AST-815) Patches:
jira_ast_815_v1.8.patch (license #5621) patch uploaded by
rmudgett (modified) ........ Merged revisions 363687 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, main/features.c: Fix recalled party B feature flags for a
failed DTMF atxfer. 1) B calls A with Dial option T 2) B DTMF
atxfer to C 3) B hangs up 4) C does not answer 5) B is called
back 6) B answers 7) B cannot initiate transfers anymore * Add
dial features datastore to recalled party B channel that is a
copy of the original party B channel's dial features datastore. *
Extracted add_features_datastore() from
add_features_datastores(). * Renamed struct ast_dial_features
features_caller and features_callee members to my_features and
peer_features respectively. These better names eliminate the need
for some explanatory comments. * Simplified code accessing the
struct ast_dial_features datastore. (closes issue ASTERISK-19383)
Reported by: lgfsantos ........ Merged revisions 363428 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, main/features.c: Hangup affected channel in error paths of
bridge_call_thread(). ........ Merged revisions 363375 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-23 16:06 +0000 [r363212] Tilghman Lesher
* /, main/astfd.c: On some platforms, O_RDONLY is not a flag to be
checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX
specification does not mandate how these 3 flags must be
specified, only that one of the three must be specified in every
call. ........ Merged revisions 363209 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-23 14:39 +0000 [r363156] Jonathan Rose
* main/manager.c, /: AST-2012-004: Fix an error that allows AMI
users to run shell commands sans authorization. As detailed in
the advisory, AMI users without write authorization for SYSTEM
class AMI actions were able to run system commands by going
through other AMI commands which did not require that
authorization. Specifically, GetVar and Status allowed users to
do this by setting their variable/s options to the SHELL or EVAL
functions. Also, within 1.8, 10, and trunk there was a similar
flaw with the Originate action that allowed users with originate
permission to run MixMonitor and supply a shell command in the
Data argument. That flaw is fixed in those versions of this
patch. (closes issue ASTERISK-17465) Reported By: David Woolley
Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose
(license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose
(license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose
(license 6182) ........ Merged revisions 363117 from
http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
Merged revisions 363141 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-23 14:07 +0000 [r363103-363107] Matthew Jordan
* /, channels/chan_sip.c: AST-2012-006: Fix crash in UPDATE
handling when no channel owner exists If Asterisk receives a SIP
UPDATE request after a call has been terminated and the channel
has been destroyed but before the SIP dialog has been destroyed,
a condition exists where a connected line update would be
attempted on a non-existing channel. This would cause Asterisk to
crash. The patch resolves this by first ensuring that the SIP
dialog has an owning channel before attempting a connected line
update. If an UPDATE request is received and no channel is
associated with the dialog, a 481 response is sent. (closes issue
ASTERISK-19770) Reported by: Thomas Arimont Tested by: Matt
Jordan Patches: ASTERISK-19278-2012-04-16.diff uploaded by Matt
Jordan (license 6283) ........ Merged revisions 363106 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* channels/chan_skinny.c: Reference skinny_subchannel object
instead of skinny_device for r363103 The check-in to resolve
ASTERISK-19592 (r363103) failed to switch to the
skinny_subchannel object instead of the skinny_device when
attempting to reference the buffer for the keypad digits. This
patch fixes that. (issue ASTERISK-19592) Reported by: Russell
Bryant
* /, channels/chan_skinny.c: AST-2012-005: Fix remotely exploitable
heap overflow in keypad button handling When handling a keypad
button message event, the received digit is placed into a fixed
length buffer that acts as a queue. When a new message event is
received, the length of that buffer is not checked before placing
the new digit on the end of the queue. The situation exists where
sufficient keypad button message events would occur that would
cause the buffer to be overrun. This patch explicitly checks that
there is sufficient room in the buffer before appending a new
digit. (closes issue ASTERISK-19592) Reported by: Russell Bryant
........ Merged revisions 363100 from
http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
Merged revisions 363102 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-21 01:45 +0000 [r362998] Richard Mudgett
* apps/app_dial.c, /: Update app_dial M and U option GOTO return
value documentation. ........ Merged revisions 362997 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-20 16:47 +0000 [r362918] Michael L. Young
* main/event.c: Add missing payload type to events API The Security
Events Framework API was changed while adding the generation of
security events in chan_sip. A payload type and name was missed
from being added to struct ie_maps. (closes issue ASTERISK-19759)
Reported by: Michael L. Young Patches: issue-asterisk-19759.diff
uploaded by Michael L. Young (license 5026)
2012-04-20 16:12 +0000 [r362816-362869] Terry Wilson
* /, main/asterisk.c: OpenBSD doesn't have rawmemchr, use strchr
(closes issue ASTERISK-19758) Reported by: Barry Miller Tested
by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller
(license 5434) ........ Merged revisions 362868 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* apps/app_speech_utils.c, /: Document Speech* apps hangup on
failure and suggest TryExec The Speech API apps return -1 on
failure, which will hang up the channel. This may not be
desirable behavior for some, but it isn't something that can be
changed without breaking people's dialplans or writing an option
to all of the Speech apps that does what TryExec already does.
This patch documents the hangup behavior of the apps, and
suggests TryExec as the solution. (closes issue AST-813) ........
Merged revisions 362815 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-19 21:59 +0000 [r362730] Walter Doekes
* funcs/func_version.c, /: Fix documentation for
${VERSION(ASTERISK_VERSION_NUM)}. ........ Merged revisions
362729 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-19 21:11 +0000 [r362681] Michael L. Young
* /, tests/test_linkedlists.c, tests/test_poll.c: Add leading and
trailing backslashes A couple of unit tests did not have have
leading or trailing backslashes when setting their test category
resulting in a warning message being displayed. Added the
backslash where needed. ........ Merged revisions 362680 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-19 21:00 +0000 [r362678] Richard Mudgett
* /, configs/queues.conf.sample: Update membermacro and membergosub
documentation in queues.conf.sample. ........ Merged revisions
362677 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-19 16:04 +0000 [r362587] Sean Bright
* /, apps/app_externalivr.c: Prevent a crash in ExternalIVR when
the 'S' command is sent first. If the first command sent from an
ExternalIVR client is an 'S' command, we were blindly removing
the first element from the play list and deferencing it, even if
it was NULL. This corrects that and also locks appropriately in
one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski
........ Merged revisions 362586 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-19 14:31 +0000 [r362537] Terry Wilson
* /, main/asterisk.c: Handle multiple commands per connection via
netconsole Asterisk would accept multiple NULL-delimited CLI
commands via the netconsole socket, but would occasionally miss a
command due to the command not being completely read into the
buffer. This patch ensures that any partial commands get moved to
the front of the read buffer, appended to, and properly sent.
(closes issue ASTERISK-18308) Review:
https://reviewboard.asterisk.org/r/1876/ ........ Merged
revisions 362536 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-19 02:27 +0000 [r362496] Matthew Jordan
* channels/chan_unistim.c, /, main/tdd.c, main/jitterbuf.c,
apps/app_sms.c, main/stdtime/localtime.c, utils/extconf.c,
addons/chan_mobile.c, main/format_pref.c, main/asterisk.c: Fix a
variety of potential buffer overflows * chan_mobile: Fixed an
overrun where the cind_state buffer (an integer array of size 16)
would be overrun due to improper bounds checking. At worst, the
buffer can be overrun by a total of 48 bytes (assuming 4-byte
integers), which would still leave it within the allocated memory
of struct hfp. This would corrupt other elements in that struct
but not necessarily cause any further issues. * app_sms: The
array imsg is of size 250, while the array (ud) that the data is
copied into is of size 160. If the size of the inbound message is
greater then 160, up to 90 bytes could be overrun in ud. This
would corrupt the user data header (array udh) adjacent to ud. *
chan_unistim: A number of invalid memmoves are corrected. These
would move data (which may or may not be valid) into the ends of
these buffers. * asterisk: ast_console_toggle_loglevel does not
check that the console log level being set is less then or equal
to the allowed log levels of 32. * format_pref: In
ast_codec_pref_prepend, if any occurrence of the specified codec
is not found, the value used to index into the array pref->order
would be one greater then the maximum size of the array. *
jitterbuf: If the element being placed into the jitter buffer
lands in the last available slot in the jitter history buffer,
the insertion sort attempts to move the last entry in the buffer
into one slot past the maximum length of the buffer. Note that
this occurred for both the min and max jitter history buffers. *
tdd: If a read from fsk_serial returns a character that is
greater then 32, an attempt to read past one of the statically
defined arrays containing the values that character maps to would
occur. * localtime: struct ast_time and tm are not the same size
- ast_time is larger, although it contains the elements of tm
within it in the same layout. Hence, when using memcpy to copy
the contents of tm into ast_time, the size of tm should be used,
as opposed to the size of ast_time. * extconf: this treats
ast_timing's minmask array as if it had a length of 48, when it
has defined the size of the array as 24. pbx.h defines minmask as
having a size of 48. (issue ASTERISK-19668) Reported by: Matt
Jordan ........ Merged revisions 362485 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-18 16:27 +0000 [r362429] Richard Mudgett
* channels/sig_pri.h, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample, /, channels/sig_pri.c: Add
ability to ignore layer 1 alarms for BRI PTMP lines. Several
telcos bring the BRI PTMP layer 1 down when the line is idle.
When layer 1 goes down, Asterisk cannot make outgoing calls.
Incoming calls could fail as well because the alarm processing is
handled by a different code path than the Q.931 messages. * Add
the layer1_presence configuration option to ignore layer 1 alarms
when the telco brings layer 1 down. This option can be configured
by span while the similar DAHDI driver teignorered=1 option is
system wide. This option unlike layer2_persistence does not
require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA
ABE-2845 ........ Merged revisions 362428 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-17 21:22 +0000 [r362360-362377] Matthew Jordan
* /, main/format_pref.c: Handle case where an unknown format is
used to get the preferred codec size In ast_codec_pref_getsize,
if an unknown format is passed to the method, no preferred codec
will be selected and a negative number will be used to index into
the format list. The method now logs an unknown format as a
warning, and returns an empty format list. (issue ASTERISK-19655)
Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1863/
* res/res_rtp_asterisk.c, /, res/res_agi.c, res/res_musiconhold.c:
Fix places in resources where a negative return value could
impact execution This patch addresses a number of modules in
resources that did not handle the negative return value from
function calls adequately. This includes: * res_agi.c: if the
result of the read function is a negative number, indicating some
failure, the result would instead be treated as the number of
bytes read. This patch now treats negative results in the same
manner as an end of file condition, with the exception that it
also logs the error code indicated by the return. *
res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor
to srcfd, and instead assigns a negative value, that file
descriptor could later be passed to functions that require a
valid file descriptor. If spawn_mp3 fails, we now immediately
retry instead of continuing in the logic. * res_rtp_asterisk.c:
if no codec can be matched between two RTP instances in a peer to
peer bridge, we immediately return instead of attempting to use
the codec payload type as an index to determine the appropriate
negotiated codec. (issue ASTERISK-19655) Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged
revisions 362362 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* main/manager.c, /, main/asterisk.c: Fix places in main where a
negative return value could impact execution This patch addresses
a number of modules in main that did not handle the negative
return value from function calls adequately, or were not
sufficiently clear that the conditions leading to improper
handling of the return values could not occur. This includes: *
asterisk.c: A negative return value from the read function would
be used directly as an index into a buffer. We now check for
success of the read function prior to using its result as an
index. * manager.c: Check for failures in mkstemp and lseek when
handling the temporary file created for processing data returned
from a CLI command in action_command. Also check that the result
of an lseek is sanitized prior to using it as the size of a
memory map to allocate. (issue ASTERISK-19655) Reported by: Matt
Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........
Merged revisions 362359 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-17 20:57 +0000 [r362357] Jonathan Rose
* res/res_config_curl.c, res/res_config_pgsql.c,
res/res_config_odbc.c, /: Make use of va_args more appropriate to
form in various res_config modules plus utils. A number of
va_copy operations weren't matched with a corresponding va_end in
res_config_odbc. Also, there was a potential for va_end to be
invoked twice on the same va_arg in utils, which would mean
invoking va_end on an undefined variable... which is bad. va_end
is removed from various functions in config_pgsql and config_curl
since they aren't making their own copy. The invokers of those
functions are responsible for calling va_end on them. (issue
ASTERISK-19451) Reported by: Walter Doekes Review:
https://reviewboard.asterisk.org/r/1848/ ........ Merged
revisions 362354 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-17 20:56 +0000 [r362305-362356] Matthew Jordan
* /, funcs/func_env.c: Fix places where a negative return from
ftello could be used as invalid input In a variety of locations
in both reading and writing a file, the result from the C library
function ftello is used as input to other functions. For the
parameters and functions in question, a negative value is invalid
input. This patch checks the return value from the ftello
function to determine if we were able to determine the current
position in the file stream and, if not, fail gracefully. (issue
ASTERISK-19655) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1863/ ........ Merged
revisions 362355 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* formats/format_g719.c, formats/format_siren7.c, /,
formats/format_sln.c, formats/format_vox.c, formats/format_wav.c,
formats/format_pcm.c, formats/format_wav_gsm.c,
formats/format_siren14.c, formats/format_gsm.c: Fix error that
caused seek format operations to set max file size to '1' or '0'
A very inappropriate placement of a ')' (introduced in r362151)
caused the maximum size of a file to be set as the result of a
comparison operation, as opposed to the result of the ftello
operation. This resulted in seeking being restricted to the
beginning of the file, or 1 byte into the file. Thanks to the
Asterisk Test Suite for properly freaking out about this on at
least one test. (issue ASTERISK-19655) Reported by: Matt Jordan
........ Merged revisions 362304 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-17 14:53 +0000 [r362264] Michael L. Young
* /, channels/chan_sip.c: Turn off warning message when bind
address is set to any. When a bind address is set to an ANY
address (udpbindport=::), a warning message is displayed stating
that "Address remapping activated in sip.conf but we're using
IPv6, which doesn't need it. Please remove 'localnet' and/or
'externaddr' settings." But if one is running dual stack, we
shouldn't be told to turn those settings off. This patch checks
if the bind address is an ANY address or not. The warning message
will now only be displayed if the bind address is NOT an ANY
address and IPv6 is being used. Also, updated the copyright year.
(closes issue ASTERISK-19456) Reported by: Michael L. Young
Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff
uploaded by Michael L. Young (license 5026) ........ Merged
revisions 362253 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-16 21:57 +0000 [r362152-362205] Matthew Jordan
* channels/chan_dahdi.c, /, channels/chan_agent.c: Fix negative
return handling in channel drivers In chan_agent, while handling
a channel indicate, the agent channel driver must obtain a lock
on both the agent channel, as well as the channel the agent
channel is using. To do so, it attempts to lock the other channel
first, then unlock the agent channel which is locked prior to
entry into the indicate handler. If this unlock fails with a
negative return value, which can occur if the object passed to
agent_indicate is an invalid ao2 object or is NULL, the return
value is passed directly to strerror, which can only accept
positive integer values. In chan_dahdi, the return value of
dahdi_get_index is used to directly index into the sub-channel
array. If dahd_get_index returns a negative value, it would use
that value to index into the array, which could cause an invalid
memory access. If dahdi_get_index returns a negative number, we
now default to SUB_REAL. (issue ASTERISK-19655) Reported by: Matt
Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........
Merged revisions 362204 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* apps/app_voicemail.c, /: Fix handling of negative return code
when storing voicemails in ODBC storage When storing a voicemail
message using an ODBC connection to a database, the voicemail
message is first stored on disk. The sound file associated with
the message is read into memory before being transmitted to the
database. When this occurs, a failure in the C library's lseek
function would cause a negative value to be passed to the mmap as
the size of the memory map to create. This would almost certainly
cause the creation of the memory map to fail, resulting in the
message being lost. (issue ASTERISK-19655) Reported by: Matt
Jordan Review: https://reviewboard.asterisk.org/r/1863 ........
Merged revisions 362201 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* formats/format_g719.c, formats/format_siren7.c,
formats/format_g729.c, formats/format_ilbc.c, /,
formats/format_sln.c, formats/format_vox.c, formats/format_wav.c,
formats/format_pcm.c, formats/format_g723.c,
formats/format_h263.c, formats/format_h264.c,
formats/format_wav_gsm.c, formats/format_siren14.c,
formats/format_gsm.c: Check for IO stream failures in various
format's truncate/seek operations For the formats that support
seek and/or truncate operations, many of the C library calls used
to determine or set the current position indicator in the file
stream were not being checked. In some situations, if an error
occurred, a negative value would be returned from the library
call. This could then be interpreted inappropriately as
positional data. This patch checks the return values from these
library calls before using them in subsequent operations. (issue
ASTERISK-19655) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1863/ ........ Merged
revisions 362151 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-13 16:04 +0000 [r362080-362084] Jonathan Rose
* apps/app_forkcdr.c, /: Make ForkCDR e option not set end time of
the newly forked CDR log Prior to this patch, ForkCDR's e option
would immediately set the end time of the forked CDR to that of
the CDR that is being terminated. This resulted in the new CDR's
end time being roughly the same as it's beginning time (which is
in turn roughly the same as the original's end time). (closes
issue ASTERISK-19164) Reported by: Steve Davies Patches:
cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012)
........ Merged revisions 362082 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, apps/app_meetme.c: Send relative path named recordings to the
meetme directory instead of sounds Prior to this patch, no effort
was made to parse the path name to determine a proper destination
for recordings of MeetMe's r option. This fixes that. Review:
https://reviewboard.asterisk.org/r/1846/ ........ Merged
revisions 362079 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-12 16:22 +0000 [r361956-361981] Kinsey Moore
* /, channels/chan_iax2.c: Make trunkfreq take effect when set
Previously, setting trunkfreq had no effect on initial load or on
reload and only ever used the default value. This causes
trunkfreq to be used appropriately on initial load and reload.
(closes issue ASTERISK-19521) Patch-by: Jaco Kroon ........
Merged revisions 361972 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* codecs/lpc10/Makefile, Makefile, build_tools/cflags.xml, /,
build_tools/menuselect-deps.in, codecs/gsm/src/k6opt.s,
configure, codecs/gsm/Makefile, configure.ac, Makefile.rules,
makeopts.in: Simplify build system architecture optimization This
change to the build system rips out any usage of PROC along with
architecture-specific optimizations in favor of using
-march=native where it is supported. This fixes broken builds on
64bit Intel systems and results in better optimized code on
systems running GCC 4.2+. Review:
https://reviewboard.asterisk.org/r/1852/ (closes issue
ASTERISK-19462) ........ Merged revisions 361955 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-11 16:07 +0000 [r361907] Jonathan Rose
* configs/queues.conf.sample, CHANGES, apps/app_queue.c: Change
default value of 'ignorebusy' on Queue members so that behavior
is more like 1.8 Prior to this patch, in order to restore that
behavior, a function would have to be used on the QueueMember to
make the ringinuse option do anything, which is pretty
unreasonable. (closes issue ASTERISK-19536) reported by: Philippe
Lindheimer Review: https://reviewboard.asterisk.org/r/1860/
2012-04-10 21:47 +0000 [r361855] Richard Mudgett
* channels/chan_dahdi.c, /: Prevent invalid access of free'd memory
if DAHDI channel during an MWI event In the MWI processing loop,
when a valid event occurs the temporary caller ID information is
deallocated. If a new DAHDI channel is successfully created, the
event is passed up to the analog_ss_thread without error and the
loop exits. If, however, the DAHDI channel is not created, then
the caller ID struct has been free'd, and the gains reset to
their previous level. This will almost certainly cause an invalid
access to the free'd memory, either in subsequent calls to
callerid_free or calls to callerid_feed. * Rework the -r361705
patch to better manage the cs and mtd allocated resources. *
Fixed use of mwimonitoractive flag to be correct if the
mwi_thread() fails to start. ........ Merged revisions 361854
from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-10 19:57 +0000 [r361658-361804] Matthew Jordan
* /, main/http.c: Fix crash caused by unloading or reloading of
res_http_post When unlinking itself from the registered HTTP
URIs, res_http_post could inadvertently free all URIs registered
with the HTTP server. This patch modifies the unregister method
to only free the URI that is actually being unregistered, as
opposed to all of them. ........ Merged revisions 361803 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* funcs/func_curl.c, /: Allow func_curl to exit gracefully if list
allocation fails during write If the global_curl_info data
structure could not be allocated, the datastore associated with
the operation would be free'd, but the function would not return.
This would later dereference the datastore, almost certainly
causing Asterisk to crash. With this patch, if the data structure
is not allocated the method will return an error code, and not
attempt any further operation. ........ Merged revisions 361753
from http://svn.asterisk.org/svn/asterisk/branches/1.8
* channels/chan_dahdi.c, /: Prevent invalid access of free'd memory
if DAHDI channel during an MWI event In the MWI processing loop,
when a valid event occurs the temporary caller ID information is
deallocated. If a new DAHDI channel is successfully created, the
event is passed up to the analog_ss_thread without error and the
loop exits. If, however, the DAHDI channel is not created, then
the caller ID struct has been free'd, and the gains reset to
their previous level. This will almost certainly cause an invalid
access to the free'd memory, either in subsequent calls to
callerid_free or calls to callerid_feed. This patch makes it so
that we only free the caller ID structure if a DAHDI channel is
successfully created, and we bump the gains back up if we fail to
make a DAHDI channel. ........ Merged revisions 361705 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, funcs/func_global.c: Change SHARED function to use a safe
traversal when modifying a variable When the SHARED function
modifies a variable, it removes it from its list of variables and
reinserts the new value at the head of the list of variables.
Doing this inside a standard list traversal can be dangerous, as
the standard list traversal does not account for the list being
changed. While the code in question should not cause a use after
free violation due to its breaking out of the loop after freeing
the variable, it could lead to a maintenance issue if the loop
was modified. This also fixes a violation reported by a static
analysis tool, which also makes this code easier to maintain in
the future. ........ Merged revisions 361657 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-06 22:00 +0000 [r361560-361607] Matthew Jordan
* /, res/res_calendar_ews.c: Fix memory leak in res_calendar_ews
when event email address node is empty If the XML calendar data
returned by a Microsoft Exchange Web Service specifies an XML
Event E-Mail Address ("EmailAddress"), and no e-mail address is
provided, a condition existed where an ast_calendar_attendee
struct would be allocated but not appended to the list of
attendees. Because of that, the memory associated with the
attendee would never be freed. This patch frees the memory if no
e-mail address is provided. ........ Merged revisions 361606 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, apps/app_meetme.c: Fix memory leak when using MeetMeAdmin 'e'
option with user specified A memory leak/reference counting leak
occurs if the MeetMeAdmin 'e' command (eject last user that
joined) is used in conjunction with a specified user. Regardless
of the command being executed, if a user is specified for the
command, MeetMeAdmin will look up that user. Because the 'e'
option kicks the last user that joined, as opposed to the one
specified, the reference to the user specified by the command
would be leaked when the user variable was assigned to the last
user that joined. ........ Merged revisions 361558 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-06 19:47 +0000 [r361522] Richard Mudgett
* main/message.c: Don't add an empty MESSAGE_DATA(key) header if it
doesn't already exist. Doing Set(MESSAGE_DATA(key)=) would add an
empty key header if the key header did not already exist. If it
already existed it would delete it. * Made msg_set_var_full()
exit early if the named variable did not already exist and the
value to set is empty.
2012-04-06 18:13 +0000 [r361472] Kinsey Moore
* channels/chan_mgcp.c, main/xmldoc.c, apps/app_voicemail.c,
res/res_clioriginate.c, channels/chan_unistim.c, main/pbx.c, /,
channels/chan_sip.c, funcs/func_strings.c,
formats/format_ogg_vorbis.c, channels/console_video.c,
channels/chan_gtalk.c, apps/app_ices.c, channels/chan_iax2.c,
res/res_config_sqlite.c, res/res_srtp.c, main/cdr.c,
main/tcptls.c, funcs/func_channel.c, channels/console_gui.c,
apps/app_sms.c, addons/chan_mobile.c, apps/app_chanspy.c: Add
missing newlines to CLI logging ........ Merged revisions 361471
from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-06 16:31 +0000 [r361422] Paul Belanger
* bridges/bridge_builtin_features.c, /, funcs/func_sysinfo.c,
bridges/bridge_multiplexed.c: Multiple revisions 361403,361412
........ r361403 | pabelanger | 2012-04-06 12:24:36 -0400 (Fri,
06 Apr 2012) | 2 lines Fix typo in svn:keywords ........ r361412
| pabelanger | 2012-04-06 12:27:30 -0400 (Fri, 06 Apr 2012) | 2
lines Fix typo in svn:keywords ........ Merged revisions
361403,361412 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-06 15:49 +0000 [r361381] Russell Bryant
* /, configs/rpt.conf.sample (removed),
configs/usbradio.conf.sample (removed), apps/rpt_flow.pdf
(removed): Remove a few more files related to chan_usbradio and
app_rpt. ........ Merged revisions 361380 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-06 14:01 +0000 [r361333] Matthew Jordan
* /, channels/chan_sip.c: Fix a typo in the warning messages for an
ignored media stream Added a '\n' to the warning messages when we
ignore a media stream due to the port number being '0'. (closes
issue ASTERISK-19646) Reported by: Badalian Vyacheslav ........
Merged revisions 361332 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-06 13:31 +0000 [r361330] Kinsey Moore
* apps/app_dial.c, /: Remove unnecessary error message in
app_dial.c The error message for failure to stop autoservice
after a gosub or macro call during a dial was removed for macro
while Asterisk 1.4 was still being actively developed. The
corresponding gosub error message was never removed. (closes
issue ASTERISK-19551) ........ Merged revisions 361329 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-05 16:53 +0000 [r361208-361270] Jonathan Rose
* /, apps/app_meetme.c: Fix MusicOnHold in MeetMe so that it always
uses the class if it's been defined There were a few instances of
restarting music on hold in meetme that would cause Asterisk to
revert to the default class of music on hold for no adequate
reason. Review: https://reviewboard.asterisk.org/r/1844/ ........
Merged revisions 361269 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, addons/ooh323cDriver.c: Fix some stuff involving calls to
memcpy and memset The important parts of the patch were already
applied through other updates. (closes issue ASTERISK-19445)
Reported by: Makoto Dei Patches: memset-memcpy-length.patch
uploaded by Makoto Dei (license 5027) ........ Merged revisions
361210 from http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, funcs/func_devstate.c: Make 'help devstate change' display
properly (get rid of excess comma) (closes issue ASTERISK-19444)
Reported by: Makoto Dei Patches:
devstate-change-usage-truncate.patch uploaded by Makoto Dei
(license 5027) ........ Merged revisions 361201 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-05-02 Asterisk Development Team
* Asterisk 10.4.0 Released.
2012-05-01 Asterisk Development Team
* Asterisk 10.4.0-rc3 Released.
* channels/chan_sip.c: Revert revision 360862
Revision 360862 was intended to improve identities sent in dialog-info
NOTIFY requests. Some users reported that hint became broken once this
was done. It's not clear exactly what part of the patch has caused
this regression, but broken hints are bad.
For now, this revision is being reverted so that the next releases of
Asterisk do not have bad behavior in them. The original reported issue
will have to be fixed differently in the next version of Asterisk.
(issue ASTERISK-16735)
2012-04-24 Asterisk Development Team
* Asterisk 10.4.0-rc2 Released.
* AST-2012-004
* AST-2012-005
* AST-2012-006
2012-04-04 Asterisk Development Team
* Asterisk 10.4.0-rc1 Released.
2012-04-04 16:38 +0000 [r361091-361143] Jonathan Rose
* main/channel.c, pbx/pbx_loopback.c, addons/chan_ooh323.c, /,
channels/chan_sip.c, main/app.c, pbx/pbx_realtime.c,
apps/app_externalivr.c, channels/chan_iax2.c,
res/res_fax_spandsp.c, apps/app_milliwatt.c: Replace GNU
old-style field designator extensions to fix clang warnings
(issue ASTERISK-19540) Reported by: Makoto Dei Patches:
clang-gnu-designator.patch uploaded by Makoto Dei (license 5027)
........ Also add from the patch the portion in res_fax_spandsp
that didn't apply to 1.8 Merged revisions 361142 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 (closes issue
ASTERISK-19540)
* /, apps/app_meetme.c: Make the MeetMeAdmin N command (mute all
nonadmins) not mute admins (Closes Issue ASTERISK-19335) Reported
by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/
........ Merged revisions 361090 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-04-03 20:08 +0000 [r360993-361041] Kinsey Moore
* /, apps/app_transfer.c: Fix the display of documentation for
Transfer This came up while fixing documentation generation for
many other cases where the argument separator was not being
displayed properly. Now that it is displayed properly, it shows
up in the wrong place for Transfer since the '/' is only required
if Tech is present. (related to issue ASTERISK-18168) ........
Merged revisions 361040 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_sip.c: Stop sending out RTCP if RTP is inactive
This change prevents Asterisk from sending RTCP receiver reports
during a remote bridge since it is no longer receiving media and
should not be reporting anything. (related to ASTERISK-19366)
........ Merged revisions 360987 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-30 21:29 +0000 [r360934] Richard Mudgett
* /, main/logger.c: Fix logger deadlock on Asterisk shutdown. The
logger_thread() had an exit path that failed to release the
logmsgs list lock. * Make logger_thread() exit path unlock the
logmsgs list lock. * Made ast_log() not queue any messages to the
logmsgs list if the close_logger_thread flag is set. (issue
ASTERISK-19463) Reported by: Matt Jordan ........ Merged
revisions 360933 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-29 23:33 +0000 [r360863-360885] Mark Michelson
* /, main/features.c: Fix potential race condition during call
pickup. Prior to this patch, a connected line update was queued
during call pickup and then an answer frame was queued. The
original caller would presumably then have his connected line
updated and then the call would be answered. In actuality, the
answer frame was not how the call ended up being answered.
Rather, an odd section in app_dial that checks if the called
channel's state is up. The result is that the order of the
connected line update and the answer were variable. In most
cases, this wasn't actually a bad thing. However, if the 'I'
option was passed to dial, the connected line update would be
inhibited. The fix is to queued the connected line after the
answer frame is queued. This way the race in app_dial is between
two conditions resulting in an answer. This way the connected
line update occurs after the answer every time. (closes issue
ASTERISK-19183) Reported by: Thomas Arimont Tested by: Thomas
Arimont Mark Michelson Patches: ASTERISK-19183.patch uploaded by
Mark Michelson (license 5049) ........ Merged revisions 360884
from http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_sip.c: Improve accuracy of identifying
information sent in dialog-info SIP NOTIFY requests. This change
makes use of connected party information in addition to caller ID
in order to populate local and remote XML elements in the
dialog-info NOTIFYs. (closes issue ASTERISK-16735) Reported by:
Maciej Krajewski Tested by: Maciej Krajewski Patches:
local_remote_hint2.diff uploaded by Mark Michelson (license 5049)
........ Merged revisions 360862 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-28 19:20 +0000 [r360717] Terry Wilson
* channels/chan_jingle.c, addons/chan_ooh323.c, /,
cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c,
channels/chan_gtalk.c: Destroy configs when they are no longer
used https://reviewboard.asterisk.org/r/1834/ ........ Merged
revisions 360712 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-27 18:23 +0000 [r360672] Mark Michelson
* /, channels/chan_sip.c: Make a debug message regarding
subscription changes more accurate. I was getting confused during
some testing why Asterisk was saying that a subscription was
being added when it was clearly being removed. This fixes that
confusion. ........ Merged revisions 360625 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-27 14:35 +0000 [r360489-360575] Jonathan Rose
* /, configure: Updates config with bootstrap where I changed
configure.ac in r360488 (issue ASTERISK-17842) Reported by: Bryon
Clark ........ Merged revisions 360574 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, configure.ac: Fix BETTER_BACKTRACES library detection for
Fedora/RedHat/CentOS (closes ASTERISK-17842) Reported by: Bryon
Clark Patches: 20110512__issue19278.diff.txt uploaded by Tilghman
Lesher (license 5003) configure_bfd_with_dl_and_iberty.patch
uploaded by Bryon Clark (license 6157) ........ Merged revisions
360488 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-26 18:41 +0000 [r360472-360476] Paul Belanger
* /, CHANGES: Update CHANGES for r360471 ........ Merged revisions
360474 from http://svn.asterisk.org/svn/asterisk/branches/1.8
* main/dnsmgr.c, /: Increase verbosity level for ast_verb messages
While this does not fix the issue of the CLI being flooded by
'doing dnsmgr_lookup' messages, increasing the verbosity level
above 5 should help minimize it. ........ Merged revisions 360471
from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-24 23:47 +0000 [r360358-360414] Russell Bryant
* funcs/func_curl.c, /: func_curl: Fix leak of an ast_str in error
handling code path. ........ Merged revisions 360413 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, apps/app_page.c: app_page: Fix a memory leak on every Page().
dial_list is a dynamically allocated array that is allocated at
the beginning of Page() based on how many devices will be dialed.
This was never being freed. ........ Merged revisions 360363 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* apps/app_jack.c, /: app_jack: fix datastore memory leak in error
handling path. ........ Merged revisions 360360 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* main/ast_expr2.c, /, main/ast_expr2.h, res/ael/ael.tab.c,
main/ast_expr2.y, main/ast_expr2f.c, res/ael/ael_lex.c,
res/ael/ael.tab.h: Multiple revisions 360356-360357 ........
r360356 | russell | 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012)
| 6 lines expression parser: Fix (theoretical) memory leak. Fix a
memory leak that is very unlikely to actually happen. If a
malloc() succeeded, but the following strdup() failed, the memory
from the original malloc() would be leaked. ........ r360357 |
russell | 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) | 6 lines
Rebuild parsers. This is needed to include the last fix to
main/ast_expr2.y. The changes look much bigger as this
regeneration of the code was done with newer versions of flex and
bison. ........ Merged revisions 360356-360357 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-24 00:37 +0000 [r360263-360310] Richard Mudgett
* main/channel.c, /, channels/sig_pri.c: Make number not available
presentation also set screening to network provided. Q.951
indicates that when the presentation indicator is "Number not
available due to interworking" for a number then the screening
indicator field should be "Network provided". * Made
ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE
when the presentation is "Number not available due to
interworking". This fix makes Asterisk consistent and it also
makes it consistent with earlier branches as far as this
presentation value is concerned. * Made pri_to_ast_presentation()
and ast_to_pri_presentation() conversions handle the "Number not
available due to interworking" case better in sig_pri.c. This
change is possible because the minimum required libpri version
(v1.4.11) has the necessary defines in libpri.h. ........ Merged
revisions 360309 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_sip.c: Add missing initialization of
update_redirecting in chan_sip.c ........ Merged revisions 360262
from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-21 14:52 +0000 [r360139] Jonathan Rose
* contrib/scripts/install_prereq, /: Update install_prereq script
to include missing GSM library for debian amd move SQLite3.
(closes issue ASTERISK-19367) Reported by: Andrew Latham Patches:
debian_install_prereq.diff uploaded by Andrew Latham (license
5985) ........ Merged revisions 360138 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-21 14:21 +0000 [r360098] Tzafrir Cohen
* /, configure, configure.ac: Also detect gmime 2.6 Also detect
gmime version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen
(License #5035) ........ Merged
revisions 360087 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-21 13:28 +0000 [r360088] Matthew Jordan
* /, channels/chan_sip.c: Ensure Asterisk sends a BYE when pending
on the final response to a re-INVITE When Asterisk detects a
hangup and cannot send a BYE due to a pending INVITE, it sets the
pendingbye flag and waits for the final response to that INVITE.
When the response is received, it transmits the BYE. If, however,
that INVITE request is a pending re-INVITE, it needs to first
send a CANCEL request to terminate the pending re-INVITE. In that
circumstance, Asterisk was, in some scenarios, clearing the
pendingbye flag after processing the CANCEL request and not
checking for a pending BYE when receiving the final 487 response
to the INVITE. This patch ensures that if the pendingbye flag is
set, it is honored regardless of the nature of the INVITE request
currently in flight. (closes issue ASTERISK-19365) Reported by:
Thomas Arimont Tested by: Thomas Arimont Patches:
bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license
6283) Review: https://reviewboard.asterisk.org/r/1807 ........
Merged revisions 360086 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-20 20:37 +0000 [r360034] Kinsey Moore
* /, apps/app_echo.c: Prevent Echo() from relaying control, null,
and modem frames Echo()'s description states that it echoes
audio, video, and DTMF except for # while it actually echoes any
frame that it receives other than DTMF #. This was causing frame
storms in the test suite in some circumstances where Echo() was
attached to both ends of a pair of local channels and control
frames were being periodically generated. Echo()'s behavior and
description have been modifed so that it only echoes media and
non-# DTMF frames. ........ Merged revisions 360033 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-20 18:11 +0000 [r359982] Sean Bright
* channels/chan_iax2.c: chan_iax2: Emit Port alongside Post in
PeerStatus AMI Event. The PeerStatus event for IAX2 channels
currently includes a header named Post which should have been
Port. So include Port along with Post when emitting the event.
We'll remove Post in trunk.
2012-03-20 17:25 +0000 [r359980] Richard Mudgett
* main/manager.c, /, include/asterisk/manager.h: Allow AMI action
callback to be reentrant. Fix AMI module reload deadlock
regression from ASTERISK-18479 when it tried to fix the race
between calling an AMI action callback and unregistering that
action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change.
Locking the ao2 object guaranteed that there were no active
callbacks that mattered when ast_manager_unregister() was called.
Unfortunately, this causes the deadlock situation. The patch
stops locking the ao2 object to allow multiple threads to invoke
the callback re-entrantly. There is no way to guarantee a module
unload will not crash because of an active callback. The code
attempts to minimize the chance with the registered flag and the
maximum 5 second delay before ast_manager_unregister() returns.
The trunk version of the patch changes the API to fix the race
condition correctly to prevent the module code from unloading
from memory while an action callback is active. * Don't hold the
lock while calling the AMI action callback. (closes issue
ASTERISK-19487) Reported by: Philippe Lindheimer Review:
https://reviewboard.asterisk.org/r/1818/ Review:
https://reviewboard.asterisk.org/r/1820/ ........ Merged
revisions 359979 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-16 20:20 +0000 [r359898] Jonathan Rose
* /, apps/app_chanspy.c: Prevent chanspy from binding to zombie
channels This patch addresses a bug with chanspy on local
channels which roughly 50% of the time would create a situation
where chanspy can latch onto a zombie channel, keeping the zombie
alive forever and causing the channel doing the spying to never
be able to hang up. (closes issue ASTERISK-19493) Reported by:
lvl Review: https://reviewboard.asterisk.org/r/1819/ ........
Merged revisions 359892 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-16 08:24 +0000 [r359810] Alec L Davis
* /, channels/sip/include/sip.h: Missed lastinvite CSeq int to
uint32_t change from Review:
https://reviewboard.asterisk.org/r/1699/ ........ Merged
revisions 359809 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-15 19:06 +0000 [r359694-359707] Matthew Jordan
* /, main/utils.c: Fix remotely exploitable stack overflow in HTTP
manager There exists a remotely exploitable stack buffer overflow
in HTTP digest authentication handling in Asterisk. The
particular method in question is only utilized by HTTP AMI. When
parsing the digest information, the length of the string is not
checked when it is copied into temporary buffers allocated on the
stack. This patch fixes this behavior by parsing out pre-defined
key/value pairs and avoiding unnecessary copies to the stack.
(closes issue ASTERISK-19542) Reported by: Russell Bryant Tested
by: Matt Jordan ........ Merged revisions 359706 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, apps/app_milliwatt.c: Fix remotely exploitable stack overrun
in Milliwatt Milliwatt is vulnerable to a remotely exploitable
stack overrun when using the 'o' option. This occurs due to the
milliwatt_generate function not accounting for
AST_FRIENDLY_OFFSET when calculating the maximum number of
samples it can put in the output buffer. This patch resolves this
issue by taking into account AST_FRIENDLY_OFFSET when determining
the maximum number of samples allowed. Note that at no point is
remote code execution possible. The data that is written into the
buffer is the pre-defined Milliwatt data, and not custom data.
(closes issue ASTERISK-19541) Reported by: Russell Bryant Tested
by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by
Russell Bryant (license 6283) Note that this patch was written by
Russell, even though Matt uploaded it ........ Merged revisions
359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
........ Merged revisions 359656 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-15 18:22 +0000 [r359620] Richard Mudgett
* apps/app_dial.c, /, apps/app_queue.c: Add missing connected line
macro calls to initial dial for Dial and Queue apps. The
connected line interception macros do not get executed when the
outgoing channel is initially created and that channel's
caller-id is implicitly imported into the incoming channel's
connected line data. If you are using the interception macros,
you would expect that they get run for every change to a
channel's connected line information outside of normal dialplan
execution. Review: https://reviewboard.asterisk.org/r/1817/
........ Merged revisions 359609 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-15 00:53 +0000 [r359454-359559] Russell Bryant
* /, channels/chan_iax2.c: chan_iax2: Fix use of uninitialized
sockaddr_in in try_transfer(). Initialize a struct sockaddr_in in
try_transfer() so that the code isn't (potentially) trying to
read from it while uninitialized. ........ Merged revisions
359558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_gtalk.c: chan_gtalk: Fix potential use of
uninitialized variable. Avoid potential use of idroster in
gtalk_alloc() before it has been initialized. ........ Merged
revisions 359508 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, apps/app_chanisavail.c: app_chanisavail: Fix use of
uninitialized variable. Ensure that status is set before it is
used by resetting it during each loop iteration. This could have
resulted in incorrect results from this app. ........ Merged
revisions 359486 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* main/udptl.c, /: udptl: Ensure fec[] in udptl_build_packet() is
initialized. Scan results indicated that this array could be used
uninitialized. At a quick look, it looks correct. In any case,
initializing it is a Good Thing (tm). ........ Merged revisions
359457 from http://svn.asterisk.org/svn/asterisk/branches/1.8
* include/asterisk/app.h, /: app.h: Always initialize
AST_DECLARE_APP_ARGS(). This patch ensures that the struct
defined by AST_DECLARE_APP_ARGS() is always fully initialized.
I'm not sure if this fixes any real bugs, but it silences a bunch
of warnings from coverity, and is generally a good thing to do
anyway. ........ Merged revisions 359452 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-14 22:28 +0000 [r359453] Richard Mudgett
* main/channel.c, /, channels/chan_agent.c,
include/asterisk/channel.h: Fix deadlock potential with some
ast_indicate/ast_indicate_data calls. Calling
ast_indicate()/ast_indicate_data() with the channel lock held can
result in a deadlock with a local channel because of how local
channels need to avoid deadlock. ........ Merged revisions 359451
from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-14 17:42 +0000 [r359358] Matthew Jordan
* /, main/jitterbuf.c: Fix incorrect jitter buffer overflow due to
missed resynchronizations When a change in time occurs, such that
the timestamps associated with frames being placed into an
adaptive jitter buffer (implemented in jitterbuf.c) are
significantly different then the previously inserted frames, the
jitter buffer checks to see if it needs to be resynched to the
new time frame. If three consecutive packets break the threshold,
the jitter buffer resynchs itself to the new timestamps. This
currently only occurs when history is calculated, and hence only
on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other
hand, are never passed to the history calculations. Because of
this, if the jump in time is greater then the maximum allowed
length of the jitter buffer, the JB_TYPE_CONTROL frames are
dropped and no resynchronization occurs. Alterntively, if the
overfill logic is not triggered, the JB_TYPE_CONTROL frame will
be placed into the buffer, but with a time reference that is not
applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger
the overflow logic until reads from the jitter buffer reach the
errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL
frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames
are unlikely to occur in multiples, it perform the
resynchronization on any JB_TYPE_CONTROL frame that breaks the
resynch threshold. Note that this only impacts chan_iax2, as
other consumers of the adaptive jitter buffer use the abstract
jitter buffer API, which does not use JB_TYPE_CONTROL frames.
Review: https://reviewboard.asterisk.org/r/1814/ (closes issue
ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt
Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw
(license 5722) ........ Merged revisions 359356 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-14 17:24 +0000 [r359355] Richard Mudgett
* apps/app_dial.c, main/channel.c, /: Fix Dial m and r options and
forked calls generating warnings for voice frames. When connected
line support was added, the wait_for_answer() variable single
changed its meaning slightly. Unfortunately, the places where
single was used did not necessarily get updated to reflect that
change. Also audio/video frames were sent to all forked calls
when the endpoints were never made compatible. * Don't pass
audio/video media frames when the channels have not been made
compatible. * Added handling of AST_CONTROL_SRCCHANGE to
app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD
because that frame can also pass a requested MOH class. (closes
issue ASTERISK-16901) Reported by: Chris Gentle (closes issue
ASTERISK-17541) Reported by: clint Review:
https://reviewboard.asterisk.org/r/1805/ ........ Merged
revisions 359344 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-14 10:54 +0000 [r359051-359260] Russell Bryant
* include/asterisk/logger.h, /, main/logger.c: Fix bogus
reads/writes of console log levels in asterisk.c This patch
updates the NUMLOGLEVELS define in logger.h to 32, to match the
fact that logger.c implements 32 log levels (because of the
custom log level stuff). asterisk.c uses this define to size an
array of levels per remote console. This array is modified in
ast_console_toggle_loglevel(), which is called by the "logger set
level" CLI command. While the documentation for the CLI command
doesn't make it terribly obvious, you can use this CLI command to
toggle a custom log level on a remote console, as well. However,
doing so led to an invalid array index in asterisk.c. This array
is read from any time a log message is written to a console. So,
all custom log level messages resulted in a bogus read if a
remote console was connected. ........ Merged revisions 359259
from http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, apps/app_externalivr.c, channels/chan_iax2.c: Fix invalid
reads/writes due to incorrect sizeof(). These few places in the
code used sizeof() on h_addr in struct hostent. This is
sizeof(char *). The correct way to get the size of this address
is to use h_length. This error would result in reads/writes of 8
bytes instead of 4 on 64-bit machines. ........ Merged revisions
359211 from http://svn.asterisk.org/svn/asterisk/branches/1.8
* main/sched.c, /: Fix inaccurate sizeof() in sched.c. This code
just needed sizeof(int), not sizeof(int *). ........ Merged
revisions 359157 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, utils/astman.c: Fix incorrect sizeof() in astman. ........
Merged revisions 359116 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, res/res_crypto.c: Fix incorrect usage of sizeof() in
res_crypto. In this case, just remove the memset(). There was a
redundant memset that is done correctly just 2 lines later.
........ Merged revisions 359110 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, res/res_adsi.c: Fix broken usage of sizeof() in res_adsi.
........ Merged revisions 359088 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, main/features.c: Fix incorrect sizeof() usage in features.c.
This didn't actually result in a bug anywhere, luckily. The only
place where the result of these memcpys was used is in app_dial,
and the only field that it read out of ast_call_feature was the
first one, which is an int, so these memcpys always copied just
enough to avoid a problem. ........ Merged revisions 359069 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, main/md5.c: Fix incorrect sizeof() on a pointer in MD5Final().
........ Merged revisions 359059 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* main/pbx.c, /: Don't use a buffer after it goes out of scope. 's'
is set to 'workspace'. Make sure 'workspace' doesn't go out of
scope while the reference to it via 's' is still used. ........
Merged revisions 359056 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* res/res_ais.c, /, res/ais/clm.c, res/ais/evt.c, res/ais/ais.h:
Dump cache of published events when a node joins the cluster.
Also use a more reliable method for stopping the poll() thread.
........ Merged revisions 359053 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* channels/chan_usbradio.c (removed), /, channels/xpmr (removed),
build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
apps/app_rpt.c (removed): Remove chan_usbradio and app_rpt. These
modules are being maintained outside of the tree and have been
for a long time now, so it doesn't make sense to keep them here.
Review: https://reviewboard.asterisk.org/r/1764/ ........ Merged
revisions 359050 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-13 20:36 +0000 [r358944-358989] Terry Wilson
* /, main/features.c: Fix setting CDR variables in the hangup
extension A previous CDR fix for setting CDR variables during a
bridge via custom dialplan features broke setting CDR variables
in the hangup extension. This patch fixes the issue. Review:
https://reviewboard.asterisk.org/r/1794/ ........ Merged
revisions 358978 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* include/asterisk/devicestate.h, /, channels/chan_sip.c,
tests/test_devicestate.c, main/devicestate.c: Make hints for
invalid SIP devices return Unavail, not idle This patch
drastically simplifies the device state aggegation code. The old
method was not only overly complex, but also made it impossible
to return AST_DEVICE_INVALID from the aggregation code. The unit
test update is as a result of fixing that bug. The SIP change
stems from a bug introduced by removing a DNS lookup for
hostname-based SIP channels. (closes issue ASTERISK-16702)
Review: https://reviewboard.asterisk.org/r/1808/ ........ Merged
revisions 358943 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-13 16:58 +0000 [r358811-358860] Tilghman Lesher
* /, UPGRADE.txt, CHANGES: Requested changes documenting the fixed
AEL functionality. ........ Merged revisions 358859 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* res/ael/pval.c, funcs/func_dialplan.c, /, tests/test_gosub.c,
utils/ael_main.c, apps/app_stack.c, utils/conf2ael.c: Enable
macros in 1.8 to find the next highest "h" extension in a
context, like in 1.4. This change restores functionality that was
present in 1.4, when AEL macros were implemented with the Macro
dialplan application. Macros are fraught with functionality
issues, because they consume a large portion of the underlying
application stack. This limits the ability of AEL users to call
many layers of subroutines, an issue which Gosub does not have
(originally tested to 100,000 levels deep). Therefore, starting
in 1.6.0, AEL macros were implemented with Gosub. However, there
were some implicit behaviors of Macro, which were not replicated
at the same time as with the transition to Gosub, one of which is
documented in the related issue. In particular, the "h" extension
is designed to execute not in the Macro context, but in the
topmost calling context. Due to legacy issues with a misapplied
bugfix many years ago, when a macro exited in 1.4, it looks in
all calling contexts, bubbling up from the deepest level until it
finds an "h" extension. Since AEL hides the complexity of the
underlying dialplan logic from the AEL programmer, it's
reasonable to assume that this behavior should not change in the
transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we
break working AEL configurations in the transition to Asterisk
1.8 LTS. This fix is the result, which implements a search for
the "h" extension in all calling Gosub contexts. Fixes
ASTERISK-19336 Patch: 20120308__ael_bugfix_for_trunk__2.diff
(License #5003) by Tilghman Lesher (with slight modifications for
1.8) Tested by: Johan Wilfer Review:
https://reviewboard.asterisk.org/r/1776/ ........ Merged
revisions 358810 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-08 16:50 +0000 [r358644] Jonathan Rose
* /, channels/chan_sip.c: Make transfer not ignore port information
with SIP. Attempting to transfer with SIP to an address like
1XXXXX@ip.ad.re.ss:5061 would fail because port would be cut from
the host string and ignored. This simply keeps chan_sip from
cutting off the port number during these kinds of transfers.
(closes issue ASTERISK-19321) Reported by: Federico Alves Review:
https://reviewboard.asterisk.org/r/1790/diff/#index_header
........ Merged revisions 358643 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-07 18:28 +0000 [r358531] Richard Mudgett
* /, channels/sig_ss7.c: Change directly setting _softhangup in
sig_ss7.c to use ast_softhangup_nolock(). Update to: (issue
ASTERISK-19372) ........ Merged revisions 358530 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-07 16:13 +0000 [r358485] Sean Bright
* /, codecs/codec_dahdi.c: Return g729 and g723.1 frames with the
number of samples set properly. If the wctc4xxp returns more than
a single packet, we need to update the number of samples in the
returned frame accordingly. Acked-by: Shaun Ruffell
........ Merged revisions 358484 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-07 15:17 +0000 [r358436-358441] Terry Wilson
* /, configs/cdr_adaptive_odbc.conf.sample: Set snarkiness = 0 in
cdr_adaptive_odbc.conf.sample ........ Merged revisions 358438
from http://svn.asterisk.org/svn/asterisk/branches/1.8
* cel/cel_odbc.c, /, cdr/cdr_adaptive_odbc.c: Add detection for
ODBC WCHAR fields Without detecting these types, cel_odbc blows
up when the character set for the table is utf8. This also wraps
cdr_adaptive_odbc's use of those types in the HAVE_ODBC_WCHAR
#ifdef seen in other parts of the code. ........ Merged revisions
358435 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-06 17:46 +0000 [r358261-358378] Richard Mudgett
* channels/chan_dahdi.c, /: Fix ring cadance setup for outgoing
calls on FXS ports. * Fix referencing the wrong variable in
chan_dahdi.c:my_set_cadence(). Thanks to Sean Bright for
compiling with -Wshadow and finding this bug. ........ Merged
revisions 358377 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/sig_ss7.c: Drop SS7 call if not connected yet when
INCOMPLETE/BUSY/CONGESTION. SS7 is a trunk protocol and should
clear a failed call as soon as possible. * Made SS7 hangup a call
immediately if it has not connected yet for
INCOMPLETE/BUSY/CONGESTION causes. Otherwise, play an appropriate
inband tone. (closes issue ASTERISK-19372) Reported by: Igor
Nikolaev ........ Merged revisions 358278 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c:
Setup DSP when SS7 call is connected or early media is available.
Outgoing SS7 calls fail to detect incoming DTMF so any bridged
channel that requires out-of-band DTMF will not work. * Added
sig_ss7_open_media() calls at appropriate places in sig_ss7.c.
The new call converts conditionaled out unconverted code and
shows that the code really did something useful. * Improved some
chan_dahdi DTMF debug messages to help track DTMF handling.
(closes issue ASTERISK-19312) Reported by: Igor Nikolaev ........
Merged revisions 358260 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-05 18:58 +0000 [r358215] Jonathan Rose
* main/manager.c, /: Eliminate double close of file descriptor in
manager.c The process_output function in manager.c attempted to
call fclose and close immediately afterwards. Since fclose
implies close, this resulted in a potential double free on file
descriptors. This patch changes that behavior and also adds error
checking to fclose and close depending on which was deemed
necessary. Also error messages. Thanks to Rosen Iliev for
pointing out the location of the problem. (closes issue
ASTERISK-18453) Reported By: Jaco Kroon Review:
https://reviewboard.asterisk.org/r/1793/ ........ Merged
revisions 358214 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-05 16:42 +0000 [r358163] Joshua Colp
* /, channels/chan_sip.c: Defer sending the connected line reinvite
if a reinvite is already in progress. (issue ASTERISK-19355)
Reported by: tomaso (closes issue AST-825) ........ Merged
revisions 358162 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-05 15:59 +0000 [r358116] Kinsey Moore
* /, channels/chan_sip.c: Ensure Asterisk acknowledges ACKs to 4xx
on Replaces errors Asterisk was not setting pendinginvite in the
upper half of handle_request_invite such that the 4xx was
retransmitted repeatedly even though an ack was received for
every retransmission. (closes issue ASTERISK-19303) Patch-by:
Jeremiah Gowdy ........ Merged revisions 358115 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-02 23:28 +0000 [r357987-358033] Terry Wilson
* channels/chan_usbradio.c, /, channels/xpmr/xpmr.c: Fix
unused-but-set-variable warnings All of these were pretty
obviously unused. Some were unused because the code that used
them was #if 0'd. In those cases, I just commented out the
unused-but-set variables. ........ Merged revisions 358029 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c,
channels/misdn/isdn_lib.c: Correct some set-but-unused variable
warnings in the mISDN library. (from kpfleming's commit to trunk
r356292) ........ Merged revisions 358011 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/xpmr/xpmr.c: Make chan_usbradio compile under dev
mode x=++x and x=x=1? Really? ........ Merged revisions 357986
from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-02 21:03 +0000 [r357941] Kinsey Moore
* /, main/ccss.c, tests/test_event.c, main/event.c,
include/asterisk/strings.h: Fix case-sensitivity for
device-specific event subscriptions and CCSS This change fixes
case-sensitivity for device-specific subscriptions such that the
technology identifier is case-insensitive while the remainder of
the device string is still case-sensitive. This should also
preserve the original case of the device string as passed in to
the event system. CCSS is the only feature affected as it is the
only consumer of device-specific event subscriptions. The second
part of this patch addresses similar case-sensitivity issues
within CCSS itself that prevented it from functioning correctly
after the fix to the events system. This adds a unit test to
verify that the event system works as expected. (closes issue
ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/
........ Merged revisions 357940 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-02 18:37 +0000 [r357895] Richard Mudgett
* main/channel.c, /, channels/sig_pri.c: Remove ISDN hold
restriction for non-bridged calls. The check if an ISDN call is
bridged before it could be placed on hold is not necessary and is
overly restrictive. The check was originally done to prevent
problems with call transfers in case a user tried to transfer a
call connected to an application to another call connected to an
application. The ISDN transfer code has not required this
restriction for quite some time because ECT could transfer any
two active calls to each other. * Remove ISDN hold restriction
for calls connected to applications. * Made
ast_waitfordigit_full() ignore AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD instead of generating a warning message.
(closes issue ASTERISK-19388) Reported by: Birger Harzenetter
Tested by: rmudgett ........ Merged revisions 357894 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-02 15:59 +0000 [r357812] Sean Bright
* /, channels/chan_iax2.c: The default value for mohinterpret is
the empty string, so when resetting to default values don't
explicitly set the value to "default." ........ Merged revisions
357811 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-02 15:50 +0000 [r357810] Richard Mudgett
* /, apps/app_chanspy.c: Fix channel reference leak in ChanSpy. *
Fix next_channel() channel reference leak in ChanSpy. (closes
issue ASTERISK-19461) Reported by: Irontec Patches:
app_chanspy_iteartor_next_unref.patch (license #6213) patch
uploaded by Irontec (issue ASTERISK-17515) ........ Merged
revisions 357809 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-02 01:05 +0000 [r357762] Mark Michelson
* main/channel.c, /: Fix race condition that can cause important
control frames (such as a hangup) to be missed. This takes two
actions. 1. Move the reading of the alertpipe in __ast_read() to
immediately before the removal of frames from the readq. This
means we won't do something silly like read from the alertpipe,
then ignore the fact that there's a frame to get from the readq
since channel's fdno is the AST_TIMING_FD. 2. When
ast_settimeout() sets the rate to 0 and the timingfunc to NULL,
if the channel's fdno is the AST_TIMING_FD, then set the fdno to
-1. This is because if the rate is 0 and the timingfunc is NULL,
it means that the channel's timing fd is being invalidated, so
any pending reads should not occur. This may actually solve more
issues than the referenced one below, but it's not known at this
time for sure. (closes issue ASTERISK-19223) reported by
Frank-Michael Wittig Review:
https://reviewboard.asterisk.org/r/1779 ........ Merged revisions
357761 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-01 14:18 +0000 [r357667] Kinsey Moore
* /, main/acl.c: Prevent outbound SIP NOTIFY packets from
displaying a port of 0 In the change from 1.6.2 to 1.8,
ast_sockaddr was introduced which changed the behavior of
ast_find_ourip such that port number was wiped out. This caused
the port in internip (which is used for Contact and Call-ID on
NOTIFYs) to be 0. This change causes ast_find_ourip to be
port-preserving again. (closes issue ASTERISK-19430) ........
Merged revisions 357665 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-29 20:39 +0000 [r357576-357620] Walter Doekes
* include/asterisk/stringfields.h, main/utils.c: Update stringfield
documentation for removed second va_list in favor of va_copy. In
r320946, the second va_list that was passed to
ast_string_field_build_va and friends, was removed. This patch
updates the documentation to reflect that.
* apps/app_dial.c, /: Fix copying of CDR(accountcode) to local
channels. In r203638, during the addition of the Channel Event
Logging, in mid-2009, this got broken in trunk and ended up in
asterisk 1.8 and higher. This fixes so the CDR(accountcode) from
the calling channel is available to dialed channels again as well
as showing up properly in the CDR's. (closes issue
ASTERISK-19384) Patches: accountcode.patch (License #6033) by
jamicque Review: https://reviewboard.asterisk.org/r/1775/
Reviewed by: Richard Mudgett ........ Merged revisions 357575
from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-28 22:29 +0000 [r357458-357497] Jonathan Rose
* /, configs/sip.conf.sample, UPGRADE-1.8.txt: Adding transport=udp
to sample sip.conf - Also changes version of Asterisk 1.8 in
UPGRADE (issue ASTERISK-19352) Reported by: jamicque Patches:
asterisk-19352-transport-warning-message-v1.patch uploaded by
Michael L. Young (license 5026) ........ Merged revisions 357490
from http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, cdr/cdr_adaptive_odbc.c: Add additional character type types
to supported data types for cdr_adaptive_odbc The reporter was
uable to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so
this patch adds those along with some other character types to
the list of types cdr_adaptive_odbc will work using the varchar
conditions. The problem wasn't really UTF8 characters as much as
it was a failure to respond to the exact type that was
declared/in use on that database. (closes issue ASTERISK-19334)
Reported By: Igor Nikolaev Patches: cdr_adaptive_odbc.patch
uploaded by Igor Nikolaev (license 6236) ........ Merged
revisions 357455 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-28 21:21 +0000 [r357421] Tilghman Lesher
* /, apps/app_stack.c: Correctly reset the dialplan priority. When
the stack frame is allocated, we save the address to which we
should return, when the Gosub returns. However, if we just want
to restore the priority, then we need to subtract 1 before
setting it. Otherwise, when a Gosub goes to a nonexistent
address, it will skip a priority in the dialplan. This is because
when we return from an application, the PBX increments the
priority for us. ........ Merged revisions 357416 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-28 20:58 +0000 [r357408] Richard Mudgett
* /, channels/sig_pri.c: Use more reasonable cause code when
rejecting incoming call waiting calls. (closes issue
ASTERISK-19397) Reported by: Birger Harzenetter Patches:
nochannel-cause.patch (license #5870) patch uploaded by Birger
Harzenetter ........ Merged revisions 357407 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-28 20:42 +0000 [r357357-357405] Jonathan Rose
* UPGRADE.txt: revision 357386 -- oops, accidentally made it 10.3
to 10.4 instead of 10.2 to 10.3 (issue ASTERISK-19352) reported
by: jamicque
* /, UPGRADE.txt, UPGRADE-1.8.txt: Moves UPGRADE.txt notes from
r357356 to a new section specific to 1.8.12 (issue
ASTERISK-19352) reported by: jamicque ........ Merged revisions
357386 from http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, UPGRADE-1.8.txt: Adds UPGRADE.txt notes to r357266 indicating
changes to transport option (issue ASTERISK-19352) Reported by:
jamicque ........ Merged revisions 357356 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-28 19:35 +0000 [r357353] Richard Mudgett
* /, apps/app_page.c: Remove dupliate 'i' option table entry in
app_page.c. (closes issue ASTERISK-19310) Reported by: Makoto Dei
Patches: app_page-duplicate-i-option.patch (license #5027) patch
uploaded by Makoto Dei ........ Merged revisions 357352 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-28 18:51 +0000 [r357318] Mark Michelson
* channels/sip/security_events.c: Add a security event for the case
where fake authentication challenge is sent.
2012-02-28 18:11 +0000 [r357271] Jonathan Rose
* /, channels/chan_sip.c: Changes transport option in sip.conf so
that using multiple instances doesn't stack. Prior to this patch,
Using "transport=" multiple times would cause them to add to one
another like allow/deny. This patch changes that behavior to
simply use the transport option specified last. Also, if no
transport option is applied now, the default will automatically
be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches:
asterisk-19352-transport-warning-message-v1.patch uploaded by
Michael L. Young (license 5026)
issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes
(license 5674) Review:
https://reviewboard.asterisk.org/r/1745/diff/#index_header
........ Merged revisions 357266 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-28 14:46 +0000 [r357213] Kevin P. Fleming
* /, Makefile.rules: Make COMPILE_DOUBLE magic actually work. The
build system has some special magic to ensure that if Asterisk is
built with --enable-dev-mode *and* DONT_OPTIMIZE, that all the
source is still compiled with the optimizer enabled (even though
the result will be thrown away), because the compiler is able to
find a great deal of coding errors and bugs as a result of
running its optimizers. Unfortunately at some point this mode got
broken, and the 'throwaway' compile of the code was no longer
done with the optimizer enabled. This patch corrects that
problem. ........ Merged revisions 357212 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-03-29 Asterisk Development Team
* Asterisk 10.3.0 Released.
2012-03-26 Asterisk Development Team
* Asterisk 10.3.0-rc3 Released.
* AST-2012-003
* AST-2012-002
* /main/manager.c, include/asterisk/manager.h: Fix AMI deadlock
regression by allowing AMI action callback to be reentrant
Fix AMI module reload deadlock from ASTERISK-18479 when it tired to
fix the race between calling an AMI action callback and
unregistering that action. Refixes ASTERISK-13874 broken by
ASTERISK-17785 change.
Locking the ao2 object guaranteed that there were no active
callbacks that mattered when ast_manager_unregister() was called.
Unfortunately, this causes the deadlock situation. The path stops
locking the ao2 object to allow multiple threads to invoke the
callback re-entrantly. There is no way to guarantee a module unload
will not crash because of an active callback. The code attempts to
minimize the chance with the registered flag and the maximum 5
second delay before ast_manager_unregister() returns.
The trunk version of the patch changes the API to fix the race
condition correctly to prevent the module code from unloading from
memory while an action callback is active.
* Don't hold the lock while calling the AMI action callback.
(closes issue ASTERISK-19487)
Reported by: Philippe Lindheimer
Review: https://reviewboard.asterisk.org/r/1818/
2012-03-06 Asterisk Development Team
* Asterisk 10.3.0-rc2 Released.
* main/acl.c: Prevent outbound SIP NOTIFY packets from displaying
a port of 0.
In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which
changed the behavior of ast_find_ourip such that port number was
wiped out. This caused the port in internip (which is used for
Contact and Call-ID on NOTIFYs) to be 0. This change causes
ast_find_ourip to be port-preserving again.
2012-01-30 22:16 +0000 [r353369-353321] Alec L Davis
* channels/sip/include/dialog.h, /, channels/chan_sip.c,
channels/sip/include/sip.h: Merged revisions 353320 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31
Jan 2012) | 18 lines RFC3261 Section 8.1.1.5. The sequence number
value MUST be expressible as a 32-bit unsigned integer * fix: use
%u instead of %d when dealing with CSeq numbers - to remove
possibility of -ve numbers. * fix: change all uses of seqno and
friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t.
Summary of CSeq numbers. An initial CSeq number must be less than
2^31 A CSeq number can increase in value up to 2^32-1 An
incrementing CSeq number must not wrap around to 0. Tested with
Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1699/ ........
* /, channels/chan_sip.c: Merged revisions 353368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r353368 | alecdavis | 2012-01-31 11:40:40 +1300 (Tue, 31 Jan
2012) | 2 lines prevent debug messsges displaying -ve Cseq
numbers. Missed in R353320 ........
2012-01-30 23:28 +0000 [r353397] Terry Wilson
* main/dnsmgr.c, /, channels/chan_sip.c, include/asterisk/dnsmgr.h:
Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr
currently takes a pointer to an ast_sockaddr and updates it
anytime an address resolves to something different. There are a
couple of issues with this. First, the ast_sockaddr is usually
the address of an ast_sockaddr inside a refcounted struct and we
never bump the refcount of those structs when using dnsmgr. This
makes it possible that a refresh could happen after the
destructor for that object is called (despite ast_dnsmgr_release
being called in that destructor). Second, the module using dnsmgr
cannot be aware of an address changing without polling for it in
the code. If an action needs to be taken on address update (like
re-linking a SIP peer in the peers_by_ip table), then polling for
this change negates many of the benefits of having dnsmgr in the
first place. This patch adds a function to the dnsmgr API that
calls an update callback instead of blindly updating the address
itself. It also moves calls to ast_dnsmgr_release outside of the
destructor functions and into cleanup functions that are called
when we no longer need the objects and increments the refcount of
the objects using dnsmgr since those objects are stored on the
ast_dnsmgr_entry struct. A helper function for returning the
proper default SIP port (non-tls vs tls) is also added and used.
This patch also incorporates changes from a patch posted by Timo
Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue
ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/
........ Merged revisions 353371 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-01-31 17:21 +0000 [r353463] Richard Mudgett
* main/manager.c, /, include/asterisk/channel.h: Fix memory leak in
error paths for action_originate(). * Fix memory leak of vars in
error paths for action_originate(). * Moved struct
fast_originate_helper tech and data members to stringfields. *
Simplified ActionID header handling for fast_originate(). * Added
doxygen note to ast_request() and ast_call() and the associated
channel callbacks that the data/addr parameters should be treated
as const char *. Review: https://reviewboard.asterisk.org/r/1690/
........ Merged revisions 353454 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-01 00:00 +0000 [r353503] Terry Wilson
* res/res_calendar.c, /: Allow res_calendar to be unloaded The
calendaring tech modules depend on res_calendar and initially
res_calendar just bumped the use count so that it couldn't be
unloaded. res_calendar can potentially create many threads and
I've seen issues where the Asterisk shutdown has failed where it
looked like these threads could be the culprit. This patch adds
unload support for res_calendar. Unloading res_calendar will also
unload the dependant tech modules as well. (closes issue
ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/
........ Merged revisions 353502 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-01 15:05 +0000 [r353551] Matthew Jordan
* /, contrib/init.d/etc_default_asterisk: Added clarification for
the VERBOSITY setting to etc_default_asterisk Clarified that
using the VERBOSITY setting in etc_default_asterisk is the same
as using the -v command line switch, which causes Asterisk to
launch in console mode. (closes issue ASTERISK-17030) Reported
by: Jonas ........ Merged revisions 353550 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-01 15:51 +0000 [r353599] Sean Bright
* /, include/asterisk/audiohook.h: Resolve an overlap in the
ast_audiohook_flags values. AST_AUDIOHOOK_TRIGGER_WRITE and
AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused
unintended side effects. This patch moves
AST_AUDIOHOOK_TRIGGER_WRITE, and updates
AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention.
This will affect existing modules that use these flags, so be
sure to recompile as necessary. (closes issue ASTERISK-19246)
Reported by: feyfre ........ Merged revisions 353598 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-01 21:16 +0000 [r353771-353721] Jonathan Rose
* /, channels/chan_sip.c: Use ast_sockaddr_stringify_fmt wrappers
for various functions in chan_sip There are a number of cleaner
looking wrappers for ast_sockaddr_stringify_fmt available which
are slightly more readable than using a direct call to
ast_sockaddr_stringify_fmt. This patch switches a number of those
calls in chan_sip to use those wrappers and is generally
harmless. (Closes issue ASTERISK-16930) Reported by: Michael L.
Young Patches: chan_sip-broken-registration-1.8.diff uploaded by
Michael L. Young (license 5026) ........ Merged revisions 353720
from http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_sip.c: Fix sip show peers port output, align
columns, and fix ami port output. A previous patch I committed
from ASTERISK-16930 unexpectedly changed some output for the AMI
action "sippeers" which this patch changes back. Also, this
aligns the output for the cli command "sip show peers" and fixes
another issue that patch introduced by using
ast_sockaddr_stringify calls multiple times without immediately
using the pointer. I also went ahead and did a little janitorial
work to clean up whitespace in _sip_show_peers. (issue
ASTERISK-16930) (closes issue ASTERISK-19281) Reported by:
Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by
Walter Doekes (license 5674) ........ Merged revisions 353769
from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-02 18:48 +0000 [r353820] Mark Michelson
* configs/http.conf.sample, main/manager.c, /, main/http.c,
configs/manager.conf.sample, include/asterisk/manager.h: Fix TLS
port binding behavior as well as reload behavior: * Removes
references to tlsbindport from http.conf.sample and
manager.conf.sample * Properly bind to port specified in
tlsbindaddr, using the default port if specified. * On a reload,
properly close socket if the service has been disabled. A note
has been added to UPGRADE.txt to indicate how ports must be set
for TLS. (closes issue ASTERISK-16959) reported by Olaf
Holthausen (closes issue ASTERISK-19201) reported by Chris
Mylonas (closes issue ASTERISK-19204) reported by Chris Mylonas
Review: https://reviewboard.asterisk.org/r/1709 ........ Merged
revisions 353770 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-02 20:11 +0000 [r353868] Richard Mudgett
* channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c:
Restore the 'w' modifier support for ISDN spans.
Dial(DAHDI/g0/1234w888) This feature also causes the sending
complete ie to be sent for switch types that do not automatically
send the ie. (EuroISDN/ETSI) The main difference between dialing
Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the
sending of the sending complete ie. (closes issue ASTERISK-19176)
Reported by: rmudgett Tested by: rmudgett ........ Merged
revisions 353867 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-02 22:27 +0000 [r353916] Kinsey Moore
* /, channels/chan_sip.c: Ensure entering T.38 passthrough does not
cause an infinite loop After R340970 Asterisk was still polling
the RTCP file descriptor after RTCP is shut down and removed. If
the descriptor happened to have data ready when the removal
occured then Asterisk would go into an infinite loop trying to
read data that it can never actually access. This change disables
the audio RTCP file descriptor for the duration of the T.38
transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan
Vrban ........ Merged revisions 353915 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-03 16:22 +0000 [r354000-353962] Jonathan Rose
* res/res_fax.c: Fixes a segfault occuring when performing attended
transfer with FAXOPT(gateway)=yes (closes issue ASTERISK-19184)
Reported by: Alexandr
* /, channels/chan_agent.c: Fixes deadlocks occuring in chan_agent
due to r335976 Bad locking order was added to chan_agent to
prevent segfaults from having no locking in a patch by irroot.
This patch addresses the bad locking order by releasing locks
before getting the right locking order to stop deadlocks from
occuring when doing multiple interactions with agents. (closes
issue ASTERISK-19285) Reported by: Alex Villacis Lasso Review:
https://reviewboard.asterisk.org/r/1708/ ........ Merged
revisions 353999 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-06 17:31 +0000 [r354217-354119] Richard Mudgett
* /, main/features.c: Add missing headers to AMI UnParkedCall event
to uniquely identify the call. The AMI UnParkedCall event was
missing the Parkinglot and Uniqueid headers that the AMI
ParkedCall event contains. (closes issue ASTERISK-19240) Reported
by: Michael Yara ........ Merged revisions 354116 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, pbx/pbx_config.c: Improved documentation of CLI "dialplan add
extension" command. * Documented dialplan add extension
,,)> format. * Allow acceptance
of command without the app-data value. There are many
applications that do no need any parameters so it is silly to
require that field for all commands. * Fixed a couple
ast_malloc/ast_free mismatches with ast_add_extension2() calls.
(closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested
by: rmudgett ........ Merged revisions 354216 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-07 15:19 +0000 [r354270] Jonathan Rose
* /, cdr/cdr_pgsql.c: Fix column duplication bug in module reload
for cdr_pgsql. Prior to this patch, attempts to reload
cdr_pgsql.so would cause the column list to keep its current data
and then add a second copy during the reload. This would cause
attempts to log the CDR to the database to fail. This patch also
cleans up some unnecessary null checks for ast_free and deals
with a few potential locking problems. (closes issue
ASTERISK-19216) Reported by: Jacek Konieczny Review:
https://reviewboard.asterisk.org/r/1711/ ........ Merged
revisions 354263 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-07 21:17 +0000 [r354349] Terry Wilson
* /, channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql:
Fix multiple SIP realtime issues 1. Set lastms to 0 when clearing
instead of "" 2. Don't set ipaddr or port to the string "(null)"
when they are empty 3. Add missing required fields, set default
for lastms to 0, and modify the length of the ipaddr field to 45
in the Postgresql realtime.sql file. (closes issue
ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/
........ Merged revisions 354348 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-09 02:25 +0000 [r354493] Russell Bryant
* main/channel.c, /: Remove some unnecessary locking from
ast_hangup(). This patch removes some unnecessary locking of the
channels container in ast_hangup(). The reason this came up is
that this lock can very quickly block the entire system. If any
of the channel cleanup code decides to block, it causes a problem
for the whole system. For example, when audiohooks get destroyed,
if that blocks for a while waiting on the mixmonitor thread to
exit because it's busy blocking on some I/O, it causes a problem
for many other threads in the meantime. Review:
https://reviewboard.asterisk.org/r/1712/ ........ Merged
revisions 354492 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-09 02:54 +0000 [r354496] Richard Mudgett
* apps/app_parkandannounce.c, /: Fix crash in ParkAndAnnounce.
Well, thats embarrasing. I forgot to initialize the caller_id
storage. (closes issue ASTERISK-19311) Reported by: tootai Tested
by: rmudgett ........ Merged revisions 354495 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-09 16:35 +0000 [r354543] Matthew Jordan
* /, channels/chan_sip.c: Fix SIP INFO DTMF handling for
non-numeric codes In ASTERISK-18924, SIP INFO DTMF handlingw as
changed to account for both lowercase alphatbetic DTMF events, as
well as uppercase alphabetic DTMF events. When this occurred, the
comparison of the character buffer containing the event code was
changed such that the buffer was first compared again '0' and '9'
to determine if it was numeric. Unfortunately, since the first
character in the buffer will typically be '1' in the case of
non-numeric event codes (10-16), this caused those codes to be
converted to a DTMF event of '1'. This patch fixes that, and
cleans up handling of both application/dtmf-relay and
application/dtmf content types. Review:
https://reviewboard.asterisk.org/r/1722/ (closes issue
ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan ........
Merged revisions 354542 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-09 17:04 +0000 [r354546] Mark Michelson
* /, res/res_fax.c: Adding reload support to res_fax.so (closes
issue ASTERISK-16712) reported by Frank DiGennaro Review:
https://reviewboard.asterisk.org/r/1713 ........ Merged revisions
354545 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-09 17:08 +0000 [r354548] Matthew Jordan
* /, channels/chan_sip.c: Clean-up of minor formatting issues in
r354542/3/4 rmudgett pointed out some formatting issues in the
check-in for ASTERISK-19290. This cleans those up. Review:
https://reviewboards.asterisk.org/r/1722/ ........ Merged
revisions 354547 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-09 19:54 +0000 [r354703-354656] Kinsey Moore
* /, main/config.c: Make the config parser remove escaping
backslashes The config parser in Asterisk does not currently
remove a backslash that is used to escape a semicolon which would
otherwise be interpreted as the start of a comment. The change
here causes that backslash to be removed, but does not create a
real escape system in the config parser. The biggest complication
with a real escape system would be breaking existing configs
everywhere (parsing \\ as \ and breaking on escaped non-semicolon
characters) even though it would be the "right" way to do things.
(closes issue ASTERISK-17121) Review:
https://reviewboard.asterisk.org/r/1724/ ........ Merged
revisions 354655 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_sip.c: Fix parsing of SIP headers where compact
and non-compact headers are mixed Change parsing of SIP headers
so that compactness of the header no longer influences which
header will be chosen. Previously, a non-compact header would be
chosen instead of a preceeding compact-form header. (closes issue
ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/
........ Merged revisions 354702 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-09 22:03 +0000 [r354750] Terry Wilson
* /, funcs/func_cdr.c: Note that CDRs are immutable once a bridge
is torn down CDRs cannot be modified after a bridge is torn down,
(e.g. after Dial() returns) even though the CDR() function may be
called. Since modifying the CDR code to change this behavior
could very easily break all kinds of things, this patch just
documents this limitation. (closes issues ASTERISK-16923) Review:
https://reviewboard.asterisk.org/r/1720/ ........ Merged
revisions 354749 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-10 18:05 +0000 [r354836] Richard Mudgett
* main/manager.c, /: Fix AMI Redirect ExtraChannel not redirecting
to the same exten and context. The astman_get_header() never
returns NULL so the check by the code for NULL would never fail.
(closes issue ASTERISK-16974) Reported by: Nuno Borges Patches:
0018325.patch (license #6116) patch uploaded by Nuno Borges
(modified) ........ Merged revisions 354835 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-10 22:00 +0000 [r354890] Jason Parker
* apps/app_voicemail.c, /: Fix a voicemail memory leak with
heard/deleted messages. open_mailbox() was changed quite a long
time ago to allocate this memory. close_mailbox() should have
been changed to be responsible for freeing it. ........ Merged
revisions 354889 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-13 16:41 +0000 [r354938] Joshua Colp
* apps/app_confbridge.c: Don't try to play sound files that do not
exist. (closes issue ASTERISK-19188) Reported by: slesru
2012-02-13 17:24 +0000 [r354959] Richard Mudgett
* res/res_config_pgsql.c, /, configs/extconfig.conf.sample: Fix
reconnecting to pgsql database after connection loss. There can
only be one database connection in res_config_pgsql just like
res_config_sqlite. If the connection is lost, the connection may
not get reestablished to the same database if the res_pgsql.conf
and extconfig.conf files are inconsistent. * Made only use the
configured database from res_pgsql.conf. * Fixed potential buffer
overwrite of last[] in config_pgsql(). (closes issue
ASTERISK-16982) Reported by: german aracil boned Review:
https://reviewboard.asterisk.org/r/1731/ ........ Merged
revisions 354953 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-13 19:51 +0000 [r355010] Joshua Colp
* /, pbx/pbx_config.c: Only allow one 'dialplan reload' to execute
at a time as otherwise they would share the same common local
context list. (closes issue AST-758) ........ Merged revisions
355009 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-13 22:03 +0000 [r355057] Richard Mudgett
* pbx/pbx_spool.c, /: Fix occasional incorrectly delayed call-file
execution. Since the dir timestamp is available at one second
resolution, we cannot know if it was updated within the same
second after we scanned it. Therefore, we will force another scan
if the dir was just modified. * Changed to force another scan if
the directory was just modified. (closes issue ASTERISK-19081)
Reported by: Knut Bakke Review:
https://reviewboard.asterisk.org/r/1688/ ........ Merged
revisions 355056 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-14 09:49 +0000 [r355137] Alexandr Anikin
* addons/chan_ooh323.c, /: call manager_event only if there is not
null channel structure (Closes issue ASTERISK-19298) Reported by:
robinfood Patches: issue19298.patch uploaded by may213 (License
#5415) ........ Merged revisions 355136 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-14 13:33 +0000 [r355183] Sean Bright
* /, channels/chan_iax2.c: Clear the high order bit from the
destination call number before sending. send_apathetic_reply
takes the incoming frame's source call number as the destination
call number for the outgoing frame. If the incoming frame was a
full frame, then the high order bit of the source call number is
set and will be interpreted as a retransmit when sent back out as
the destination call number. ........ Merged revisions 355182
from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-14 15:53 +0000 [r355229] Jason Parker
* /, configs/cdr_sqlite3_custom.conf.sample: Don't enable sqlite3
CDRs by default in sample configs. ........ Merged revisions
355228 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-14 16:27 +0000 [r355271] Mark Michelson
* /, channels/chan_sip.c: Properly invert the return of a strncmp
call. This was causing identification that should have been made
private to be public. (closes issue AST-814) reported by Patrick
Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson
(license 5430) ........ Merged revisions 355268 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-14 18:14 +0000 [r355375-355320] Richard Mudgett
* /, cel/cel_sqlite3_custom.c: Fix lock typo that should be unlock
in cel_sqlite_custom reload. (closes issue ASTERISK-19356)
Reported by: Alex Villacis Lasso Patches:
asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch
(license #5617) patch uploaded by Alex Villacis Lasso Review:
https://reviewboard.asterisk.org/r/1740/ ........ Merged
revisions 355319 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
formats/format_ogg_vorbis.c: Fix voicemail problems when using
ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file
format because it did not implement the seek and tell format
callbacks among other problems. Since we were already using the
libvorbis and libvorbisenc libraries we can use libvorbisfile as
it is also part of the vorbis library package. * Made use the
libvorbisfile to handle the ogg/vorbis file stream. The
format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.
(closes issue ASTERISK-16926) Reported by: sque Patches:
ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded
by sque ........ Merged revisions 355365 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-15 17:25 +0000 [r355530-355449] Sean Bright
* /, channels/chan_iax2.c: Use TRUNK_CALL_START as originally
intended. Back in r646, TRUNK_CALL_START was added and defined as
0x4000. That same value was also hard-coded in one part of the
IAX2 code instead of using the #define. TRUNK_CALL_START has
changed over the years (for dealing with LOW_MEMORY), but the
hard-coded usage was never updated to match. This patch fixes
that. ........ Merged revisions 355448 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_iax2.c: Only use maxtrunkcall and
maxnontrunkcall in chan_iax2 if IAX_OLD_FIND is specified. These
variables are only accessed from the IAX_OLD_FIND path, so there
is no reason to keep them updated otherwise. ........ Merged
revisions 355458 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_iax2.c: When IAX2 debugging is enabled, make
sure to log 'apathetic' messages too. ........ Merged revisions
355529 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-16 18:32 +0000 [r355620-355575] Richard Mudgett
* /, res/res_monitor.c: Fix AMI Monitor action without File header
converting channel name into filename. * Fix potential Solaris
crash if Monitor application has a urlbase and no fname_base
option. ........ Merged revisions 355574 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, configure, include/asterisk/autoconfig.h.in,
autoconf/ast_c_declare_check.m4 (added), configure.ac,
formats/format_ogg_vorbis.c: Fix compile problem when old version
of libvorbisfile v1.1.2 is used. The principle difference between
libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition
of the predefined callbacks OV_CALLBACKS_xxx in
vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the
configure script to detect if libvorbisfile.h declares
OV_CALLBACKS_NOCLOSE. * Copied the declaration of
OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile.
(closes issue ASTERISK-19370) Reported by: Jonn Taylor ........
Merged revisions 355608 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-16 20:01 +0000 [r355623] Sean Bright
* /, main/audiohook.c: Revert a change to
audio_audiohook_write_list that had no affect. When I made this
change initially, I was under the false impression that the
audiohooks structure remained on the channel after all of the
hooks had been detached. This is not the case, ast ast_read takes
care of removing the audiohooks structure if the lists are empty.
........ Merged revisions 355622 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-17 19:06 +0000 [r355733] Mark Michelson
* /, channels/chan_sip.c: Fix regressions with regards to route-set
creation on early dialogs. This fixes two main issues: 1.
Asterisk would send a CANCEL to the route created by the
provisional response instead of using the same destination it did
in the initial INVITE. 2. If a new route set arrives in a 200 OK
than was in the 1XX response (perfectly possible if our outbound
INVITE gets forked), then the route set in the 200 OK needs to
overwrite the route set in the 1XX response. (closes issue
ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten
Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson
(license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt
(license 6034) Review: https://reviewboard.asterisk.org/r/1749
........ Merged revisions 355732 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-17 19:34 +0000 [r355794-355747] Sean Bright
* /, channels/chan_iax2.c: Pass the correct value to
ast_timer_set_rate() for IAX2 trunking. IAX2 uses the trunkfreq
variable to determine how often to send trunk packets, but this
value is in milliseconds while ast_timer_set_rate() expects the
rate argument to be ticks per second. So we divide 1000 by
trunkfreq and pass that in instead. With a default of 20ms, this
change makes IAX2 send trunk packets every 20ms instead of every
50ms. Tracked down by myself and Bob Wienholt. ........ Merged
revisions 355746 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* configs/iax.conf.sample, /, channels/chan_iax2.c: Don't allow
trunkfreq to be greater than 1000ms. ........ Merged revisions
355793 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-18 07:58 +0000 [r355851] Alec L Davis
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_ss7.h, /, channels/sig_analog.h, channels/sig_pri.c,
channels/sig_ss7.c: push 'outgoing' flag from sig_XXX up to
chan_dahdi 'p->outgoing' in chan_dahdi and sig_analog wern't kept
in sync, particulary FXS ast_hangup didn't clear the 'outgoing'
flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync.
Now provides a callback for all the low level sig_XXX modules.
(issue ASTERISK-19316) alecdavis (license 585) Reported by:
Jeremy Pepper Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1747/ ........ Merged
revisions 355850 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-18 17:02 +0000 [r355896-355895] Paul Belanger
* /: Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)
........ Merged revisions 355839 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /: Revert commit
2012-02-19 17:50 +0000 [r355998-355902] Sean Bright
* /, channels/chan_iax2.c: Set the length of the ast_sockaddr, so
that we can set it's port later. Without this, the call to
ast_sockaddr_set_port a few lines later is a noop. ........
Merged revisions 355901 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_iax2.c: Add some boilerplate documentation for
IAXVAR and IAXPEER. ........ Merged revisions 355904 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* channels/chan_dahdi.c, /: Change some debug messages from
LOG_DEBUG to ast_debug. ........ Merged revisions 355949 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* channels/chan_dahdi.c, /: This was a LOG_NOTICE, so roll it back.
........ Merged revisions 355952 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, channels/chan_iax2.c: Remove spurious warning when
'qualifyfreqnotok' is set successfully. (closes issue
ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright
Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John
Covert (license 5512) ........ Merged revisions 355997 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-21 04:30 +0000 [r356074] Kinsey Moore
* main/ccss.c: Add missing newline to ccss state change
notification Move along, nothing to see here...
2012-02-21 11:17 +0000 [r356108] Sean Bright
* /, channels/chan_iax2.c: Make 'iax2 show callnumber usage' output
make sense when an IP is passed in. ........ Merged revisions
356107 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-22 14:53 +0000 [r356215] Matthew Jordan
* /, channels/chan_sip.c: Merged revisions 356214 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012)
| 27 lines Fix potential buffer overrun and memory leak when
executing "sip show peers" The "sip show peers" command uses a
fix sized array to sort the current peers in the peers
ao2_container. The size of the array is based on the current
number of peers in the container. However, once the size of the
array is determined, the number of peers in the container can
change, as the peers container is not locked. This could cause a
buffer overrun when populating the array, if peers were added to
the container after the array was created. Additionally, a memory
leak of the allocated array would occur if a user caused the
_show_peers method to return CLI_SHOWUSAGE. We now create a
snapshot of the current peers using an ao2_callback with the
OBJ_MULTIPLE flag. This size of the array is set to the number of
peers that the iterator will iterate over; hence, if peers are
added or removed from the peers container it will not affect the
execution of the "sip show peers" command. Review:
https://reviewboard.asterisk.org/r/1738/ (closes issue
ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas
Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey
Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan
(license 6283) ........
2012-02-22 21:18 +0000 [r356297] Terry Wilson
* main/loader.c, res/res_calendar.c, /,
include/asterisk/calendar.h: Track module use count for
res_calendar If the res_calendar module was followed immediately
by one of the calendar tech modules and "core stop gracefully"
was run, Asterisk would crash. This patch adds use count tracking
for res_calendar so that it is unloaded after the tech modules
when shutting down gracefully. It is now not possible to unload
all the of the calendar modules via "module unload
res_calednar.so", but it is still possible to unload them all via
"module unload -h res_calendar.so". Review:
https://reviewboard.asterisk.org/r/1752/ ........ Merged
revisions 356291 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-23 03:23 +0000 [r356431-356428] Paul Belanger
* /, apps/app_rpt.c: Multiple revisions 356290,356335,356337
........ r356290 | pabelanger | 2012-02-22 15:20:29 -0500 (Wed,
22 Feb 2012) | 4 lines Fix -Werror=unused-but-set-variable
compiler error (gcc 4.6.2) Review:
https://reviewboard.asterisk.org/r/1763/ ........ r356335 |
pabelanger | 2012-02-22 16:29:25 -0500 (Wed, 22 Feb 2012) | 2
lines Add back strsep() function for previous commit ........
r356337 | pabelanger | 2012-02-22 16:36:37 -0500 (Wed, 22 Feb
2012) | 2 lines Missed one strsep() function ........ Merged
revisions 356290,356335,356337 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* addons/chan_ooh323.c, /: Fix -Werror=unused-but-set-variable
compiler error (gcc 4.6.2) ........ Merged revisions 356430 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-23 15:40 +0000 [r356476] Mark Michelson
* /, channels/chan_sip.c: Fix ACK routing for non-2xx responses.
When we send an ACK for a 2xx response to an INVITE, we are
supposed to use the learned route set. However, when we receive a
non-2xx final response to an INVITE, we are supposed to send the
ACK to the same place we initially sent the INVITE. We had been
doing this up until the changes went in that would build a route
set from provisional responses. That introduced a regression
where we would use the learned route set under all circumstances.
With this change, we now will set the destination of our ACK
based on the invitestate. If it is INV_COMPLETED then that means
that we have received a non-2xx final response (INV_TERMINATED
indicates a 2xx response was received). If it is INV_CANCELLED,
then that means the call is being canceled, which means that we
should be ACKing a 487 response. The other change introduced here
is setting the invitestate to INV_CONFIRMED when we send an ACK
*after* the reqprep instead of before. This way, we can tell in
reqprep more easily what the invitestate is prior to sending the
ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer
patches: ASTERISK-19389v2.patch uploaded by Mark Michelson
(license #5049) (with some slight modifications prior to commit)
........ Merged revisions 356475 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-02-23 19:52 +0000 [r356522] Richard Mudgett